The research on the Novelty Detection System (NDS) (called as VENUS) at the authors' universities has generated exciting results. For example, we can detect an abnormal behavior (such as cars thefts from the parking lot) from a series of video frames based on the cognitively motivated theory of habituation. In this paper, we would like to describe the implementation strategies of lower layer protocols for using large-scale Wireless Sensor Networks (WSN) to NDS with Quality-of-Service (QoS) support. Wireless data collection framework, consisting of small and low-power sensor nodes, provides an alternative mechanism to observe the physical world, by using various types of sensing capabilities that include images (and even videos using Panoptos), sound and basic physical measurements such as temperature. We do not want to lose any 'data query command' packets (in the downstream direction: sink-to-sensors) or have any bit-errors in them since they are so important to the whole sensor network. In the upstream direction (sensors-to-sink), we may tolerate the loss of some sensing data packets. But the 'interested' sensing flow should be assigned a higher priority in terms of multi-hop path choice, network bandwidth allocation, and sensing data packet generation frequency (we hope to generate more sensing data packet for that novel event in the specified network area).
The focus of this paper is to investigate MAC-level Quality of Service (QoS) issue in Wireless Sensor Networks (WSN) for Novelty Detection applications. Although QoS has been widely studied in other types of networks including wired Internet, general ad hoc networks and mobile cellular networks, we argue that QoS in WSN has its own characteristics. In wired Internet, the main QoS parameters include delay, jitter and bandwidth. In mobile cellular networks, two most common QoS metrics are: handoff call dropping probability and new call blocking probability. Since the main task of WSN is to detect and report events, the most important QoS parameters should include sensing data packet transmission reliability, lifetime extension degree from sensor sleeping control, event detection latency, congestion reduction level through removal of redundant sensing data. In this paper, we will focus on the following bi-directional QoS topics: (1) Downstream (sink-to-sensor) QoS: Reliable data query command forwarding to particular sensor(s). In other words, we do not want to lose the query command packets; (2) Upstream (sensor-to-sink) QoS: transmission of sensed data with priority control. The more interested data that can help in novelty detection should be transmitted on an optimal path with higher reliability. We propose the use of Differentiated Data Collection. Due to the large-scale nature and resource constraints of typical wireless sensor networks, such as limited energy, small memory (typically RAM < 4K bytes) and short communication range, the above problems become even more challenging. Besides QoS support issue, we will also describe our low-energy Sensing Data Transmission network Architecture. Our research results show the scalability and energy-efficiency of our proposed WSN QoS schemes.
The interaction between multicasting and real-time multimedia streams poses various new and interesting problems in research on communication protocols and architectures. For example, display systems of heterogeneous users in the same session may have different or even varying latency, display resolution, and/or processing capabilities. In this paper, we propose a receiver-oriented resource reservation mechanism, called Dynamic Management of QoS with Priority (DMQP), for multimedia multicasting over the Internet that provides QoS guarantees with service differentiation for heterogeneous users. In DMQP a real-time application requests QoS by specifying a range of bandwidth values and delay, and the network tries to reserve resources for it within its bandwidth range. The service differentiation is achieved by classifying the end-users into normal user and prioritized user. When the number of end-users increases, the new end-users are not simply rejected. Instead, all nodes, including receiver node(s), sender node(s) and intermediate node(s), readjust their reserved resources dynamically to admit more end-users, as long as their minimum bandwidth requirement is met. By treating the bandwidth requirements as ranges, DMQP provides the flexibility needed for operation in the dynamic Internet environment. It has the significant benefit of allowing more flexible sharing of available resources among applications. We have conducted simulations to evaluate the performance of the proposed mechanism.
The integration of telemedicine with medical micro sensor technology (Mobile Sensor Networks for Telemedicine applications -- MSNT) provides a promising approach to improve the quality of people's lives. This type of network can truly implement the goal of providing health-care services anytime and anywhere. Our research in this field generates the following outcomes that are reported in this paper: (1) We propose a mobile sensor network infrastructure to support the third-generation telemedicine applications; (2) An energy-efficient query resolution mechanism in large-scale mobile sensor networks is used for critical medical data collections; (3) To provide the guaranteed mobile <i>QoS</i> for arriving multimedia calls, a new multi-class call admission control mechanism is proposed which is based on dynamically forming a reservation pool for handoff requests. We used discrete-event-based simulation model using OPNET to verify our scheme. The simulation results show that our system can satisfy the adaptive QoS requirements in large-scale telemedicine sensor networks.
In this paper, we present a low-complexity RVLC decoding scheme for MPEG-4 video (including the effect of DC/AC prediction) that recovers more blocks and sometimes more MBs from error propagation region of corrupted video packets, as compared to the MPEG-4 scheme. The remaining blocks and MBs are concealed, by using maximally smooth error concealment scheme. It is shown that the proposed scheme achieves better data recovery, both in terms of PSNR and perceptual quality. In addition, we present more conditions for better error detection than those suggested in MPEG-4, and also discuss properties of error propagation in corrupted video packets. Since the schemes are purely decoder based, the compliance with the standard is fully maintained.
This paper introduces the applications of image processing and JPEG2000 in pathology. Our study aims to provide pathologists valuable assistance on examining biopsy samples of prostate and support their decision. The digital color microscopic image is processed only within the regions of interest instead of overall processing. In this way, some diagnostic information is enhanced to provide better visual effect. JPEG2000 is used for the storage and interactive transmission of the large pathological images. Our simulation results show that the use of JPEG2000 can significantly save storage space without compromising the diagnostic value of the image.
In this paper we describe a Monte Carlo simulation for time resolved fluorescence. In the past information on steady state measurements have been reported. However we feel that a lot more information and insight could be gained by the use of time resolved fluorescence spectroscopy. We have developed a Monte Carlo simulation to study the fluorescence signal generated by fluorophores distributed in a scattering medium. The simulation uses a semi-infinite medium with a thickness of 1cm. We have used the simulation to study the effect of the change in optical properties of the medium on the TPSF (temporal point spread function) generated. We have also investigated the effect of the increased radial separation of the detector on the TPSF. We have observed a shift in the Tmax (time at which the peak intensity is reached) in accordance with diffusion theory.
We wanted to validate our simulation by seeing how well we could derive the optical properties of the medium from the TPSF produced from simulation. We fitted the TPSF to an adjusted form of the diffusion theory to find scattering coefficient, μ<sub>s</sub>, and we have used an analytical model of time resolved fluorescence to extract the absorption coefficient, μ<sub>a</sub>. The results obtained were better than previously reported.
Many investigations in the literature have studied the migration of photons in fluorescent and non-fluorescent applications, both in the steady-state and time-resolved, within a tissue-simulating medium. However, none have addressed the specific issue of quantified the subsequent migration path of the emitted photons. This is important since fluorescent spectroscopy has been gaining acceptance as an important diagnostic tool. In this paper we show the migration patterns of fluorescent photons with respect to time in a scattering medium such as tissue. The images produced are of the paths of the emission photons that reach the detector. We are able to observe how they migrate at different times. We investigate the effect of the absorption and scattering coefficients on the migration patterns, and how it could give us information about methods to detect inhomogeneities in the medium. The images produced give us information about the accuracy of the estimation of the optical properties in particular medium. Finally, we study how the detector position and absorption and scattering coefficients affect the source of the photons that reach the detector. We discover that during the rising time of the temporal spectrum most of the detected fluorescent photons are being generated very close to the source and the path followed by these photons are localized near the detector. Therefore, we could explore this for the best orientation and location of the detector and source positions to locate inhomogenieties and also source of fluorescence photons within the medium.
Quality of Service (QoS) is an important issue in the next generation wireless networks providing multimedia services. In this paper, we address the connection-level QoS provisioning in wireless multimedia networks, measured by the connection blocking and dropping probabilities. The connection-level QoS for multimedia services are guaranteed by achieving the minimum connection blocking probability subject to the constraint on the handoff dropping probability. A dynamic call admission control scheme is proposed to provide connection-level QoS in wireless multimedia networks. This scheme adopts a novel strategy called prompt-decreasing/timer-increasing (PDTI) to dynamically adjust the threshold for handoff channel reservation. It can maintain the handoff dropping probability at a target rate predefined in the system specification, while maximizing resource utilization and minimizing the new call blocking rate. The proposed solution is a measurement-based method that is practical for real-world deployment. Simulations are carried out to prove the efficiency of the proposed PDTI scheme.
The Data Over Cable Service Interface Specifications (DOCSIS) of the Multimedia Cable Network System (MCNS) organization intends to support IP traffics over HFC (hybrid fiber/coax) networks with significantly higher data rates than analog modems and Integrated Service Digital Network (ISDN) links. The availability of high speed-access enables the delivery of high quality audio, video and interactive services. To support quality-of-service (QoS) for such applications, it is important for HFC networks to provide effective medium access and traffic scheduling mechanisms. In this work, a novel scheduling mechanism and a new bandwidth allocation scheme are proposed to support multimedia traffic over DOCSIS (Data Over Cable System Interface Specification)-compliant cable networks. The primary goal of our research is to improve the transmission of real-time variable bit rate (VBR) traffic in terms of throughput and delay under DOCSIS. To support integrated services, we also consider the transmission of constant bit rate (CBR) traffic and non-real-time traffic in the simulation. To demonstrate the performance, we compare the result of the proposed scheme with that of a simple multiple priority scheme. It is shown via simulation that the proposed method provides a significant amount of improvement over existing QoS scheduling services in DOCSIS. Finally, a discrete-time Markov model is used to analyze the performance of the voice traffic over DOCSIS-supported cable networks.
A novel scheduling mechanism and a new bandwidth allocation scheme are proposed in this work to support multimedia traffic over DOCSIS (Data Over Cable System Interface Specification)-compliant cable networks. The primary goal of our research is to improve the transmission of real-time variable bit rate (VBR) traffic in terms of throughput and delay under the current DOCSIS specifications. To support integrated services, we also consider the transmission of constant bit rate (CBR) traffic and non-real-time traffic in simulation. To demonstrate the performance, we compare the result of the proposed scheme with that of a simple multiple priority scheme. It is shown via simulation that the proposed method provides a significant amount of improvement over existing DOCSIS QoS scheduling services.
Proc. SPIE. 4671, Visual Communications and Image Processing 2002
KEYWORDS: Code division multiplexing, Detection and tracking algorithms, Data modeling, Video, Control systems, Telecommunications, Multimedia, Systems modeling, Standards development, Global system for mobile communications
A call admission control (CAC) scheme and a resource-reservation estimation (RRE) method suitable for the interference-based wireless system, such as wide-band code division multiple access (W-CDMA), are proposed in this work. The proposed CAC scheme gives preferential treatment to high priority handoff calls by pre-reserving a certain amount of interference margin called the interference guard margin (IGM). The amount of guard margin is determined by the measurement performed by the RRE module in base stations. Each RRE module dynamically adjusts the level of guard margin by considering traffic conditions in neighboring cells based upon handoff requests. A service model is adopted to support multiple services, which includes mobile terminal's data rate, different levels of priorities, mobility and rate adaptivity characteristics. Simulations are conducted with OPNET to study the performance of the proposed scheme in terms of the objective function, blocking probabilities and system utilization under different traffic conditions.
Proc. SPIE. 4529, Enabling Technologies for 3G and Beyond
KEYWORDS: Code division multiplexing, Video, Control systems, Computer simulations, Telecommunications, Multimedia, Performance modeling, Broadband telecommunications, Systems modeling, Global system for mobile communications
A call admission control (CAC) scheme and a resource reservation estimation (RRE) method suitable for the wide-band code division multiple access (W-CDMA) systems are proposed in this work. The proposed CAC scheme gives preferential treatment to high priority calls, such as handoff calls, by pre-reserving a certain amount of channel margin against the interference effect. It is called the interference guard margin (IGM) scheme. The amount of guard margin is determined by the measurement performed by the RRE module in base stations. Each RRE module dynamically adjusts the level of guard margin by referencing traffic conditions in neighboring cells based upon users' requests. A comprehensive service model is adopted to accommodate the scenario of multiple services supported in the W-CDMA system. The service model of consideration includes not only mobile terminal's service rate (source rate) but also different levels of priorities, mobility and rate adaptivity characteristics. Simulations are conducted with OPNET to study the performance of the proposed scheme in term of the objective function under different traffic conditions.
In this research, we first address the QoS issue in different levels of wireless multimedia networks, and present a generic QoS framework to meet the requirements of different applications and services adaptively. Then, we focus on the connection-level QoS, measured by the connection blocking and dropping probabilities. A service model consisting of three service classes designed for connection-level QoS provisioning is proposed. The underlying network of consideration employs different call admission control and resource reservation schemes to allocate resources adaptively to each service class according to their haracteristics and requirements. The system is analyzed by a multi-dimensional model. Simulations are conducted based on the model analysis to evaluate the system performance.
The Data Over Cable Service Interface Specifications (DOCSIS) of the Multimedia Cable Network System (MCNS) organization intends to support IP traffics over HFC (hybrid fiber/coax) networks with significantly higher data rates than analog modems and Integrated Service Digital Network (ISDN) links. The availability of high speed-access enables the delivery of high quality audio, video and interactive services. To support quality-of-service (QoS) for such multimedia applications, it is important for HFC networks to provide effective medium access and traffic scheduling mechanisms. In this work, we consider an HFC network that has a shared upstream channel for transmissions from stations assigned with different service priorities to the headend. We first present a multilevel priority collision resolution scheme with adaptive contention window adjustment. The proposed collision resolution scheme separates and resolves collisions for different classes of critically delay-sensitive and best effort traffics, thereby, achieving the capability for preemptive priorities. To enhance the performance of the proposed scheme, we adopt a novel methodology in which the headend dynamically selects the optimal backoff window size according to the estimate of the number of contending stations for each priority class. A traffic scheduling policy with multiple priority queues is also employed in the headend to schedule data transmissions. This scheduling strategy is used to satisfy bandwidth requirements for higher priority traffics. Simulations are conducted by using OPNET. We present a set of simulation scenarios to demonstrate the performance efficiency of the proposed scheme.
Proc. SPIE. 4209, Multimedia Systems and Applications III
KEYWORDS: Mathematical modeling, Signal to noise ratio, Control systems, Telecommunications, Multimedia, Wireless communications, Signal detection, Systems modeling, Radio propagation, Mobile communications
A dynamic call admission control (CAC) and its associated resource
reservation (RR) schemes are proposed in this research based on the
guard channel (GC) concept for a wireless cellular system supporting
multiple QoS classes. A comprehensive service model is developed, which
includes not only mobile terminals' bandwidth requirements but also
their different levels of priorities, rate adaptivity and mobility. The
proposed CAC policy selects the resource access thresold according to
the estimated number of incoming call requests of different QoS classes.
The amount of resources to be reserved is dynamically adjusted by
considering neighboring-cell higher-priority calls which are likely to
handoff. The handoff interaction between adjacent cells is estimated by
using radio propagation in terms of the signal-to-noise ratio (SNR) and
the distance of each active call in neighboring cells. Experiments are
conducted by using the OPNET simulator to study the performance of the
proposed scheme under various traffic conditions. It is shown that
better QoS guarantees can be provided by the proposed CAC and RR
Proc. SPIE. 4209, Multimedia Systems and Applications III
KEYWORDS: Mathematical modeling, Networks, Telecommunications, Multimedia, Wireless communications, Network architectures, Model-based design, Standards development, Global system for mobile communications, Mobile communications
An adaptive system to support multimedia applications in a wireless
network environment is investigated in this research. The proposed
system is hierarchical in nature with a cluster of mobile end-hosts
connected to a base station, and base stations are connected to a
supervisory node, which is in turn connected to a wired infrastructure.
Based on our previous experience, the system is designed with an
improved service model and a novel call admission scheme that enables
more efficient radio channel usage. The service model takes into
account both user mobility and the nature of multimedia traffic.
Specifically, delay- and/or rate-adaptive services are explored to
increase the network performance and utilization under highly variable
network connectivity and transmission capacity. The new call admission
control scheme is developed under a hybrid call admission control
framework. Together with an adaptive resource reservation scheme, it
can be adapted to the network situation as well as application
requirements. Simulation studies are conducted to show the performance
improvement of the proposed scheme.
EBCOT, as the baseline algorithm of JPEG-2000 final draft, is an efficient image coding technique. It is inherently more error resilient than many other wavelet-based schemes due to its independent coding of blocks in each subband. However, the loss of data of a block, in any lower frequency subband, in EBCOT can still degrade the perceptual image quality considerably. As robust entropy codes are used, the information of image content can help recover damaged regions of blocks. This paper discusses the use of reversible variable length codes (RVLC) and data partitioning for coding lower frequency subbands in EBCOT, instead of arithmetic codes. The selection of RVLC is based on content of images, which is classified as active and shape. We have observed that the proposed approach has very little additional bit-rate overhead and improved performance in the presence of errors from the help of content.
One key issue of providing multimedia services over a mobile wireless network is the quality of service (QoS) support in the presence of changing network connectivity. The trend of using pico-cells in wireless networks to gain more spatial efficiency increases the rate of call handoffs when mobile users move from one cell to another. Frequent handoffs make it very difficult to support QoS for multimedia applications. In this research, we investigate a potential solution to meet the challenge of seamless resource transition during frequent handoffs by combining the differentiated QoS service model and the priority handoff mechanism. We perform simulations with OPNET. Results show a tradeoff between system utilization and handoff blocking rates for different QoS classes.
The provisioning of quality of service (QoS) in future wireless communication networks is a complex problem due to the presence of changing network connectivity, user mobility, and shared, noisy, highly variable and limited communication links. In this paper, we propose a QoS management framework in wireless multimedia networks. A new comprehensive service model considering both traffic characteristics and user mobility for wireless multimedia networks is proposed. Based on this proposed service model, adaptive resource scheduling, admission control and resource reservation schemes are applied appropriately. Simulation results show that the proposed scheme can achieve higher network utilization and the resulting multimedia traffic can get better quality of service guarantees at different levels.
Existing techniques for coding a general 3D graphic model are very sensitive to errors. The coded bitstream can be easily ruined with even a single bit error in the topology structure part. As a result, the reconstruction of the original mesh becomes difficult or even impossible. In this research, we propose a new approach for error resilient coding of 3D graphic models, and reconstruct the original model to the maximum extent from corrupted 3D mesh data while maintaining a satisfactory compression performance. In the proposed scheme, the 3D mesh is first divided into a number of small pieces by using any standard mesh segmentation algorithm. A new structure, i.e. the joint boundary, is then retrieved from those divided pieces. It is used to stitch segmented pieces back to form the original model. We develop a new coding algorithm for the joint boundary. The coded joint boundary topology and the first 3 bit-planes of coded joint boundary geometry are protected using the forward error correction (FEC) code. They can be decoded free of error. Consequently, they provide the anchor-vertices and anchor-links in the 3D space, and form the key structure in error detection, recovery and concealment of corresponding pieces. It is demonstrated that the proposed coding scheme can achieve very good results for the bit error rate (BER) less than 10<SUP>-3</SUP> while maintaining high coding efficiency.
An error resilient coding method for 3D graphic models is proposed in this research, which exploits the topology and geometry information of the original mesh to convert the mesh to a pre- defined 3D structure via morphing, and then partitions the morphed structure into a set of smaller pieces via volume splitting. By using the morphing and splitting processes, the error propagation length is greatly reduced since it is confined to each individual piece rather than the whole mesh. Different pieces are connected via their joint boundaries. By coding the joint boundaries separately, individual pieces can be reconstructed independently of each other. Consequently, error resilient coding can be conveniently applied to each piece. The number of pieces of partitioning is determined by channel error rates. It is demonstrated that the proposed method can achieve satisfactory results in an environment of bit error rates less than 10<SUP>-3</SUP>.
An integrated system for the robust coding of topological mesh data of arbitrary 3D graphic models is investigated in this work. The proposed system can achieve higher error resiliency with a low bit-rate overhead. This system mainly consists of two major modules, i.e. the segmentation module and the reversible variable length coding (RVLC) module. The segmentation module is used to divide an arbitrary 3D mesh into a group of smaller, uniform and independent segments depending on the error rate. Errors introduced in network transmission can be limited to the current segment instead of the whole mesh, which reduces the error propagation length drastically. The reversible variable length coding module is applied to each individual segment. It allows the recovery of a large portion of data from a corrupted segment due to the two-way decoding capability of RVLC. The amount of retransmitted data can thus be greatly reduced. In this research, two specific types of RVLC are considered, their parameters are carefully selected to match the symbol probability distributions. Experimental results show that an average overhead of 10 - 20% is required by the proposed scheme in comparison with the original error-free coding technique for the 300 testing 3D graphic models to given an excellent performance in the presence of noise.
EBCOT is an efficient image coding technique, which divides each subband into independently coded blocks. It is therefore inherently more error resilient than many other wavelet-based schemes. However, the loss of data of a block in any lower frequency subband in EBCOT can still degrade the perceptual image quality considerably. This paper discusses the use of reversible variable length codes (RVLC) and data partitioning for coding lower frequency subbands in EBCOT, instead of the use of arithmetic codes. RVLCs are known to have superior error recovery properties due to their two-way decode capability. It is demonstrated that the proposed approach has very little additional bit-rate overhead and significantly improved the performance in the presence of errors.
An improved self-synchronizing Huffman code is proposed to decrease the error propagation length in the compressed bit stream caused by errors during transmission. After regaining synchronization, the decoder may still not be able to align each symbol with its correct location due to the wrong number of previously decoded symbols in the error propagation region. A scheme to identify the probable error location and then move symbols towards their correct positions is also proposed by exploiting the correlation between coefficients of the current subband and their parent subband. Experiments under different error conditions are performed, and it is demonstrated that the proposed error resilient techniques provide more a robust codec with very little sacrifice in the coding efficiency.
A visual pattern-based image compression technique is presented, in which 4 X 4 image blocks are classified in perceptually significant `shade' and `edge' classes. The proposed technique attempts to make use of neighboring blocks to encode a shade or an edge block by exploiting the Human Visual System characteristics. To reduce correlation present in the shade regions of an image, the mean intensity of a shade block is predicted from the neighboring shade blocks, and the error mean is computed. The error mean of a block is then encoded by choosing an appropriate quantizer based on its predicted mean. The quantizer has been designed after a careful study of the distribution of the error mean of shade blocks in test images, based on Weber's law, to maximize the compression ratio without introducing any visible error. Higher dimension shade blocks (8 X 8 and 16 X 16) are also formed, by merging adjacent shade blocks which further reduces the inter-block correlation. An edge block is assumed to contain two uniform intensity regions (low and high intensity) separated by a transition region. Hence, an edge block can be encoded by coding its edge pattern, low or high intensity and gradient. In order to reduce the inter-block correlation, the edge pattern and mean intensity (low or high) are predicted. The mean intensity of error is encoded by using an appropriate quantizer. Therefore, this technique achieves higher compression ratios, as compared to other visual pattern- based techniques, at very low computational complexity.