ppxscal/embedding_network huggingface dataset:
Green's Function Estimation in Homogeneous and Scattering Medium for Time Reversal Signal Processing (TRSP) Communication
Authors: Quy Nguyen Dinh
Venue: Proceedings of the Second International Conference on Digital Telecommunications (2007)

Time Reversal Signal Processing (TRSP) is a new approach to mitigate the multiple path (multipath) interference in reverberation environment. TRSP has smaller computation resource than classical equalization since TRSP uses phase conjugation technique for compensating distortion. Thus, TRSP technique is applied in receiver for enhancing transmitted code information sequence in communication systems in a highly reverberant environment. The Time Reversal (TR) operation can be performed either on transmission or reception using estimated Green's function that is spatio-temporal channel response. Thus, Green's function estimation is the first step to set up TR communication. In this paper, Green's function estimation is proposed in homogeneous and scattering medium to establish the feasibility of TRSP in communication. The proposed system can estimate Green's function with minimal computation and better accuracy.
Pseudo-organ boundary conditions applied to a computational fluid dynamics model of the human aorta
Authors: Joong Yull Park, Chan Young Park, Chang Mo Hwang, Kyung Sun, Byoung Goo Min
Venue: Computers in Biology and Medicine (2007)

In three-dimensional numerical studies of the aorta, it is difficult to apply proper boundary conditions at the end of each major aortic branch because of interactions between blood and organs. Organs and body parts were assumed to be likened to cylindrically shaped porous media, so-called pseudo-organs, and treated in the computational domain as forms of hemodynamic resistance. Permeability functions were determined from two-dimensional axisymmetric computations of each aortic branch and these functions were then used in an unsteady three-dimensional simulation of the complete aorta. Substantially accurate cardiac output (5.91L/min) and blood distributions to the major branches were predicted.
Trace: parallel trace replay with approximate causal events
Authors: Michael P. Mesnier, Matthew Wachs, Raja R. Sambasivan, Julio Lopez, James Hendricks, Gregory R. Ganger, David O'Hallaron
Venue: Proceedings of the 5th USENIX conference on File and Storage Technologies (2007)

//TRACE is a new approach for extracting and replaying traces of parallel applications to recreate their I/O behavior. Its tracing engine automatically discovers internode data dependencies and inter-I/O compute times for each node (process) in an application. This information is reflected in per-node annotated I/O traces. Such annotation allows a parallel replayer to closely mimic the behavior of a traced application across a variety of storage systems. When compared to other replay mechanisms, //TRACE offers significant gains in replay accuracy. Overall, the average replay error for the parallel applications evaluated in this paper is below 6%.
A new model of generating chaos
Authors: Heng Song, Ji-zhang Zuo, Feng-ming Zhang
Venue: Proceedings of the Fourth IEEE International Conference on Cognitive Informatics (2005)

According to the problem of being difficult to control randomicity of chaos series generated by neutral networks (NN), this paper presents a new model of generating chaos based on adaptive neurofuzzy, inference system (ANFIS) whose inside parameters are adjusted by delaminating-adaptation genetic algorithm. Simulation shows this new model can generate chaos series with excellent randomicity. And randomicity of chaos series can be intelligently controlled by delaminating-adaptation genetic algorithm. The new model is suitable for spread spectrum communications.
Genetic Algorithm Based Approach for Hierarchical SOC Test Scheduling
Authors: Chandan Giri, Dilip Kumar Reddy Tipparthi, Santanu Chattopadhyay
Venue: Proceedings of the International Conference on Computing: Theory and Applications (2007)

In today's system-on-chip (SOC) design process heterogeneous technology cores are integrated at several layers of hierarchy. Hence, multilevel test access mechanism (TAM) optimization is necessary for modular testing of hierarchical SOCs that contain earlier generation SOCs as embedded megacores. Unlike previous works that mostly assumes flat test hierarchy, the proposed technique considers the design hierarchy constraints for the cores in case of non-interactive design transfer between core vendor and core user. The proposed SOC test scheduling technique is used to minimize the test application time for the hierarchical SOCs. Experimental results are presented for three ITC'02 benchmark circuits containing megacores and results shown 55% improvement over recently proposed methods for hierarchical SOC test scheduling.
The SDA Model: A Set Theory Approach
Authors: David Riano
Venue: Proceedings of the Twentieth IEEE International Symposium on Computer-Based Medical Systems (2007)

Procedural knowledge in medicine is embedded in Clinical Practice Guidelines whose textual condition makes it difficult to share and to reuse. Several languages for formal definition of clinical practice guidelines have been proposed to overcome these difficulties. In order to deal with the huge amount of medical situations, these languages use to be extensive and complex in such a way that they, and the knowledge they are used to represent, are arduous to understand and to manage by non-trained general practitioners. The SDA* model is introduced as an alternative language that promotes representation capability and simplicity in such a way that not only computers, but also health care professionals are able to understand and manage easily without any sort of training. Here, a description of this model from a set theory perspective is provided.
Cross-Community Collaboration and Learning in Customer-Driven Software Engineering Student Projects
Authors: Birgit Krogstie, Bendik Bygstad
Venue: Proceedings of the 20th Conference on Software Engineering Education & Training (2007)

This paper explores collaboration and learning between stakeholders in customer-driven student projects. The research objectives are to obtain empirically based knowledge on how students relate to stakeholders in customer-driven projects, and to suggest implications for the pedagogical design of the project courses. Empirical data was collected from two Bachelor courses in software engineering at two learning institutions in Norway. To make sense of the interaction between the three stakeholders in the project: the student groups, the university and the customer, we build on Wenger's concept of communities of practice and on the concept of boundary objects. Our analysis highlights that students, through practical experience in the projects, learn to balance the requirements and expectations from different stakeholders in designing a working technical solution - a valuable skill for software engineers. We argue that for students to learn to balance stakeholders' interests in the best possible way, visibility of stakeholders' goals should be focused throughout the projects. Explicit reference to the goals should be incorporated into project artifacts serving as boundary objects. Collaboration technologies providing standard shared workspace functionality are seen as adequate to support this.
Signal processing techniques for efficient use of transmit diversity in wireless communications
Authors: G. W. Wornell
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1996. on Conference Proceedings., 1996 IEEE International Conference - Volume 02 (1996)

A class of efficient strategies for exploiting transmit antenna diversity on fading channels is developed. These techniques, which we refer to as linear antenna precoding, asymptotically transform a nonselective Rayleigh fading channel into a nonfading, simple white marginally Gaussian noise channel with no intersymbol interference. Linear antenna precoding requires no additional power or bandwidth, and is also attractive in terms of computation, robustness and delay considerations.
Reinforcing the understanding of signal processing concepts using audio exercises
Authors: J. W. Pierre
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1999. on 1999 IEEE International Conference - Volume 06 (1999)

In the near future, multimedia techniques will be used more extensively in signal processing education because the technology is available and the benefits to student learning and information retention are high. Using a variety of teaching techniques helps a wider range of students, who have different learning styles, and enhances student skills in their weaker areas. This paper describes a number of audio signal processing homework exercises used to reinforce concepts of signal processing. These exercises include some fundamental concepts of DSP (quantization, aliasing, Fourier analysis, and filtering) and more advanced areas (sampling rate conversion, LCMV filtering, and adaptive filtering). All these exercises use the signal processing and audio capabilities of MATLAB. A Web page for these homework exercises is being developed.
Partly hidden Markov model and its application to speech recognition
Authors: T. Iobayashi
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1999. on 1999 IEEE International Conference - Volume 01 (1999)

A new pattern matching method, the partly hidden Markov model, is proposed and applied to speech recognition. The hidden Markov model, which is widely used for speech recognition, can deal with only piecewise stationary stochastic process. We solved this problem by introducing the modified second order Markov model, in which the first state is hidden and the second one is observable. In this model, not only the feature parameter observations but also the state transitions are dependent on the previous feature observation. Therefore, even the complicated transient can be modeled precisely. Some simulation experiments showed the high potential of the proposed model. From the results of the word recognition test is was observed that the error rate was reduced by 39% compared with the normal HMM.
Visualizing Evolving Searches with EvoBerry
Authors: Edward Suvanaphen, Jonathan C. Roberts
Venue: Proceedings of the 11th International Conference Information Visualization (2007)

Studies show that roughly one-third of searches that are performed on the web require the user to initiate subsequent searches. Bates [1] theorized that with every search the user will encounter new information, which in turn leads to new ideas and directions. This process causes a change, not simply in the query terms being used but also to the nature of the information retrieval task itself; Bates called this the Evolving Search. She also noted that Evolving Searches utilize many different information sources, generate substantial quantities of data and require easy methods to save and recall data. Although current search tools are exceptionally efficient at locating highly ranked pages, the tools do not encourage or support the user in an evolving search. In this paper we present techniques that aid users to find, view and manage data produced from their evolving searches. In particular, we introduce the EvoBerry environment, which we have developed for use with evolving searches. EvoBerry includes methods to visualize additional search result information (such as length of page or file type), manage the user's session and browsing history, compare result sets, and store and bookmark items for future reference.
Audio coding with a dynamic wavelet packet decomposition based on frequency-varying modulated lapped transforms
Authors: M. Purat
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1996. on Conference Proceedings., 1996 IEEE International Conference - Volume 02 (1996)

Optimum time-frequency decompositions are very useful in audio coding applications, because the signal energy can be maximally concentrated even for the wide variety of audio signal characteristics. Moreover, this signal representation is particularly well suited for a perceptual weighting of the quantization noise. The well known tree structure of cascaded 2-channel filterbanks allows a very flexible optimization, leading to a signal adaptive, dynamic wavelet packet decomposition. A major drawback of this technique are strong spectral side lobes which produce clearly audible aliasing in perceptual coders. We present a new dynamic wavelet packet decomposition, based on modulated lapped transforms, which allows the same flexibility while avoiding the disadvantage mentioned above. We propose a scheme for low bit rate audio coding that efficiently exploits the high energy concentration. This new codec yields excellent audio quality at about 55 kb/s for monophonic signals.
Adaptive delay concealment for Internet voice applications with packet based time-scale modification
Authors: Fang Liu
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 2001. on IEEE International Conference - Volume 03 (2001)

Much effort has been involved in packet-level error control and delay jitter concealment for Internet audio applications. A packet-based time-scale modification scheme for speech signals is developed to provide adaptive delay concealment, at the receiver for an Internet voice session in this work. The adaptive playout algorithm strives to minimize packet droppings for late-arrival packets and premature packets while maintaining the end-to-end delay constraint. By stretching the length of voice segments and incorporating silence intervals, the proposed algorithm is able to accommodate fluctuating delays including delay spikes quickly. It is verified by experiments that the proposed adaptive playout algorithm improves the received speech intelligence under a tightly bounded average playout delay.
Natural language generation using an information-slim representation
Authors: Daniel Marcu
Venue:  (2006)

In this dissertation, I propose a new natural language generation paradigm, based on direct transformation of textual information into well-formed textual output. I support this language generation paradigm with theoretical contributions in the field of formal languages, new algorithms, empirical results, and software implementations. At the core of this work is a novel representation formalism for probability distributions over finite languages. Due to its convenient representation and computational properties, this formalism supports a wide range of language generation needs, from sentence realization to text planning. Based on this formalism, I describe, implement, and analyze theoretically a family of algorithms that perform language generation using direct transformations of text. These algorithms use stochastic models of language to drive the generation process. I perform extensive empirical evaluations using my implementation of these algorithms. These evaluations show state-of-the-art performance in automatic translation, and significant improvements in state-of-the-art performance in abstractive headline generation and coherent discourse generation.
Robust constant modulus arrays based on fractional lower-order statistics
Authors: M. Rupi
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1999. on 1999 IEEE International Conference - Volume 05 (1999)

This paper addresses the problem of blind equalization for digital communications using an array of sensors at the receiver to copy constant modulus signals in the presence of heavy-tailed additive channel noise. First, we demonstrate the negative effects of channel noise to the original constant modulus algorithm (CMA) cost function in terms of convergence. Then, we introduce a new CMA criterion based on the fractional lower-order statistics (FLOS) of the received data. The proposed criterion is able to mitigate impulsive noise at the receiver and at the same time restores the constant modulus character of the transmitted communication signal. We perform an analytical study of the properties of the new cost function and we illustrate its convergence behavior through computer simulations.
Design, distribution and management of object-oriented software
Authors: Arindam Banerji, David Cohn, Dinesh Kulkarni
Venue: Proceedings of the USENIX Applications Development Symposium Proceedings on USENIX Applications Development Symposium Proceedings (1994)

The promise of object-oriented software has been somewhat dimmed by the continuing need for source code familiarity to realize the goals of code-reuse and manageability. Software design has been hampered by the infeasibility of predicting all possible circumstances of use. Composing applications out of reusable components has remained a myth, primarily due to limitations of the simplistic shared library model. This paper proposes a three-pronged attack on these limitations of object-oriented software in the context of C++. A flexibility framework which facilitates the extension and modification of software without recompilations or source-code familiarity is described. Partially resolved loadable subclasses that may be distributed as reusable units for type-safe application composition. Specific programming guidelines which allow implementors to create software that may be fine-tuned at run-time according to application characteristics is described. Thus, the paper proposes a set of tools, techniques and guidelines that can facilitate the construction of application frameworks.
Controllability-driven Power Virus Generation for Digital Circuits
Authors: K. Najeeb, Karthik Gururaj, V. Kamakoti, Vivekanand M. Vedula
Venue: Proceedings of the 20th International Conference on VLSI Design held jointly with 6th International Conference: Embedded Systems (2007)

The problem of peak power estimation in CMOS circuits is essential for analyzing the reliability and performance of circuits at extreme conditions. The Power Virus problem involves finding input vectors that cause maximum dynamic power dissipation (maximum toggles) in circuits. In this paper, an approach for power virus generation for both combinational and sequential circuits is presented. The basic intuition behind the approach is to use the 0- and 1- controllability measures of the gate outputs in the circuit to guide the D-Algorithm. The proposed technique was employed on the ISCAS'85 and ISCAS'89 circuits. The results of the above show a significant increase in power dissipation when compared to the best known existing techniques reported in the literature.
Recognition of elderly speech and voice-driven document retrieval
Authors: S. Anderson
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 1999. on 1999 IEEE International Conference - Volume 01 (1999)

We have collected a corpus of 78 hours of speech from 297 elderly speakers, with an average age of 79. We find that acoustic models built from elderly speech provide much better recognition than do non-elderly models (42.1 vs. 54.6% WER). We also find that elderly men have substantially higher word error rates than elderly women (typically 14% absolute). We report on other experiments with this corpus, dividing the speakers by age, by gender, and by regional accent. Using the resulting "elderly acoustic model", we built a document-retrieval program that can be operated by voice or typing. After usability tests with 110 speakers, we tested the final system on 37 elderly speakers. Each retrieved 4 documents from a database of 86,190 Boston Globe articles, 2 by typing and 2 by speech. We measured how quickly they retrieved each article, and how much help they required. We find no difference between spoken and typed queries in either retrieval times or in amount of help required, regardless of age, gender, or computer experience. However, users perceive speech to be substantially faster, and overwhelmingly prefer speech to typing.
Linear regression under maximum a posteriori criterion with Markov random field prior
Authors: Xintian Wu
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 2000. on IEEE International Conference - Volume 02 (2000)

Speaker adaptation using linear transformations under the maximum a posteriori (MAP) criterion has been studied in this paper. The purpose is to improve the matrix estimation in the widely used maximum likelihood linear regression (MLLR) adaptation, which might generate poorly structured transform matrices when adaptation data are sparse. Unlike traditional MAP based adaptations, many known prior distributions of HMM parameters, such as normal-Washart priors, do not have a close form solution in the transform estimation. In Markov random field linear regression (MRFLR), the prior distribution of HMM parameters is modeled by Markov random field, which leads to a close form solution of estimating the linear transforms. Experimental results show that MRFLR outperforms MLLR when adaptation data are sparse, and converges to the MLLR performances when more adaptation data are available.
Passive acoustic source localization for video camera steering
Authors: Yiteng Huang
Venue: Proceedings of the Acoustics, Speech, and Signal Processing, 2000. on IEEE International Conference - Volume 02 (2000)

A multi-input one-step least-squares (OSLS) algorithm for passive source localization is proposed. It is shown that the OSLS algorithm is mathematically equivalent to the so-called spherical interpolation (SI) method but with less computational complexity. The OSLS/SI method uses spherical equations (instead of hyperbolic equations) and solves them in a least-squares sense. Based on the adaptive eigenvalue decomposition time delay estimation method previously proposed by the same authors and the OSLS source localization algorithm, a real-time passive source localization system for video camera steering is presented. The system demonstrates many desirable features such as accuracy, portability, and robustness.
Towards a Video QoE Definition in Converged Networks
Authors: Mukundan Venkataraman, Shamik Sengupta, Mainak Chatterjee, Raja Neogi
Venue: Proceedings of the Second International Conference on Digital Telecommunications (2007)

Most resource allocation decisions in converged multi-service networks are presently driven by QoS parameters and service level agreements. Network operators are realizing that such policy decisions ought to be based on Quality of Experience (QoE), which is a better measure of subjective video perception. Video QoE has two components: (a) Zapping and (b) Video Quality or Video Mean Opinion Score (MOS). Zapping anomalies are well understood and easily characterized, but there is no consensus on what the definition of a Video-MOS should be. This is largely because there is little understanding of qualitative and quantitative subjective degradations caused by networking events like loss, delay, jitter, and error. In this work we conduct extensive experiments on a simulation testbed in a search for a good Video-MOS definition, with specific emphasis on quality degradations due to network transmission. We deploy "collector" nodes that can gather various statistics of a given flow. We also identify key parameters that such collector nodes must maintain such that an overlay of such nodes can identity impairment points. Lightweight Video- MOS definition evolving out this work can be used in collector overlay networks to solve problems like root-cause analysis, capacity-planning and various network optimization problems.