String Matching Problems with Parallel Approaches An Evaluation for the Most Recent Studies

In recent years string matching plays a functional role in many application like information retrieval, gene analysis, pattern recognition, linguistics, bioinformatics etc. For understanding the functional requirements of string matching algorithms, we surveyed the real time parallel string matching patterns to handle the current trends. Primarily, in this paper, we focus on present developments of parallel string matching, and the central ideas of the algorithms and their complexities. We present the performance of the different algorithms and their effectiveness. Finally this analysis helps the researchers to develop the better techniques

Interim research assessment 2003-2005 - Computer Science

This report primarily serves as a source of information for the 2007 Interim Research Assessment Committee for Computer Science at the three technical universities in the Netherlands. The report also provides information for others interested in our research activities

Error tolerant multimedia stream processing: There's plenty of room at the top (of the system stack)

There is a growing realization that the expected fault rates and energy dissipation stemming from increases in CMOS integration will lead to the abandonment of traditional system reliability in favor of approaches that offer reliability to hardware-induced errors across the application, runtime support, architecture, device and integrated-circuit (IC) layers. Commercial stakeholders of multimedia stream processing (MSP) applications, such as information retrieval, stream mining systems, and high-throughput image and video processing systems already feel the strain of inadequate system-level scaling and robustness under the always-increasing user demand. While such applications can tolerate certain imprecision in their results, today's MSP systems do not support a systematic way to exploit this aspect for cross-layer system resilience. However, research is currently emerging that attempts to utilize the error-tolerant nature of MSP applications for this purpose. This is achieved by modifications to all layers of the system stack, from algorithms and software to the architecture and device layer, and even the IC digital logic synthesis itself. Unlike conventional processing that aims for worst-case performance and accuracy guarantees, error-tolerant MSP attempts to provide guarantees for the expected performance and accuracy. In this paper we review recent advances in this field from an MSP and a system (layer-by-layer) perspective, and attempt to foresee some of the components of future cross-layer error-tolerant system design that may influence the multimedia and the general computing landscape within the next ten years. © 1999-2012 IEEE

A Single Chip System for Sensor Data Fusion Based on a Drift-diffusion Model

Current multisensory system face data communication overhead in integrating disparate sensor data to build a coherent and accurate global phenomenon. We present here a novel hardware and software co-design platform for a heterogeneous data fusion solution based on a perceptual decision making approach (the drift-diffusion model). It provides a convenient infrastructure for sensor data acquisition and data integration and only uses a single chip Xilinx ZYNQ-7000 XC7Z020 AP SOC. A case study of controlling the moving speed of a single ground-based robot, according to physiological states of the operator based on heart rates, is conducted and demonstrates the possibility of integrated sensor data fusion architecture. The results of our DDM-based data integration shows a better correlation coefficient with the raw ECG signal compare with a simply piecewise approach

An Intelligent Architecture Based on Field Programmable Gate Arrays Designed to Detect Moving Objects by Using Principal Component Analysis

This paper presents a complete implementation of the Principal Component Analysis (PCA) algorithm in Field Programmable Gate Array (FPGA) devices applied to high rate background segmentation of images. The classical sequential execution of different parts of the PCA algorithm has been parallelized. This parallelization has led to the specific development and implementation in hardware of the different stages of PCA, such as computation of the correlation matrix, matrix diagonalization using the Jacobi method and subspace projections of images. On the application side, the paper presents a motion detection algorithm, also entirely implemented on the FPGA, and based on the developed PCA core. This consists of dynamically thresholding the differences between the input image and the one obtained by expressing the input image using the PCA linear subspace previously obtained as a background model. The proposal achieves a high ratio of processed images (up to 120 frames per second) and high quality segmentation results, with a completely embedded and reliable hardware architecture based on commercial CMOS sensors and FPGA devices

CABE : a cloud-based acoustic beamforming emulator for FPGA-based sound source localization

Microphone arrays are gaining in popularity thanks to the availability of low-cost microphones. Applications including sonar, binaural hearing aid devices, acoustic indoor localization techniques and speech recognition are proposed by several research groups and companies. In most of the available implementations, the microphones utilized are assumed to offer an ideal response in a given frequency domain. Several toolboxes and software can be used to obtain a theoretical response of a microphone array with a given beamforming algorithm. However, a tool facilitating the design of a microphone array taking into account the non-ideal characteristics could not be found. Moreover, generating packages facilitating the implementation on Field Programmable Gate Arrays has, to our knowledge, not been carried out yet. Visualizing the responses in 2D and 3D also poses an engineering challenge. To alleviate these shortcomings, a scalable Cloud-based Acoustic Beamforming Emulator (CABE) is proposed. The non-ideal characteristics of microphones are considered during the computations and results are validated with acoustic data captured from microphones. It is also possible to generate hardware description language packages containing delay tables facilitating the implementation of Delay-and-Sum beamformers in embedded hardware. Truncation error analysis can also be carried out for fixed-point signal processing. The effects of disabling a given group of microphones within the microphone array can also be calculated. Results and packages can be visualized with a dedicated client application. Users can create and configure several parameters of an emulation, including sound source placement, the shape of the microphone array and the required signal processing flow. Depending on the user configuration, 2D and 3D graphs showing the beamforming results, waterfall diagrams and performance metrics can be generated by the client application. The emulations are also validated with captured data from existing microphone arrays.</jats:p

Design of Pattern Matching Systems: Pattern, Algorithm, and Scanner

Pattern matching is at the core of many computational problems, e.g., search engine, data mining, network security and information retrieval. In this dissertation, we target at the more complex patterns of regular expression and time series, and proposed a general modular structure, named character class with constraint repetition (CCR), as the building block for the pattern matching algorithm. An exact matching algorithm named MIN-MAX is developed to support overlapped matching of CCR based regexps, and an approximate matching algorithm named Elastic Matching Algorithm is designed to support overlapped matching of CCR based time series, i.e., music melody. Both algorithms are parallelized to run on FPGA to achieve high performance, and the FPGA-based scanners are designed as a modular architecture which is parameterizable and can be reconfigured by simple memory writes, achieving a perfect balance between performance and deployment time

Efficient audio signal processing for embedded systems

We investigated two design strategies that would allow us to efficiently process audio signals on embedded systems such as mobile phones and portable electronics. In the first strategy, we exploit properties of the human auditory system to process audio signals. We designed a sound enhancement algorithm to make piezoelectric loudspeakers sound "richer" and "fuller," using a combination of bass extension and dynamic range compression. We also developed an audio energy reduction algorithm for loudspeaker power management by suppressing signal energy below the masking threshold. In the second strategy, we use low-power analog circuits to process the signal before digitizing it. We designed an analog front-end for sound detection and implemented it on a field programmable analog array (FPAA). The sound classifier front-end can be used in a wide range of applications because programmable floating-gate transistors are employed to store classifier weights. Moreover, we incorporated a feature selection algorithm to simplify the analog front-end. A machine learning algorithm AdaBoost is used to select the most relevant features for a particular sound detection application. We also designed the circuits to implement the AdaBoost-based analog classifier.PhDCommittee Chair: Anderson, David; Committee Member: Hasler, Jennifer; Committee Member: Hunt, William; Committee Member: Lanterman, Aaron; Committee Member: Minch, Bradle