The excessive memory access required to perform motion compensation when decoding compressed video is one of the main limitations in improving the performance of an H.264/AVC decoder. This paper proposes an H.264/AVC decoder that employs three techniques to reduce external memory access events: efficient distribution of reference frame data, on-chip cache memory, and frame memory recompression. The distribution of reference frame data is optimized to reduce the number of row activations during SDRAM access. The novel cache organization is proposed to simplify tag comparisons and ease the access to consecutive 4×4 blocks. A recompression algorithm is modified to improve compression efficiency by using unused storage space in neighboring blocks as well as the correlation with the neighboring pixels stored in the cache. Experimental results show that the three techniques together reduce external memory access time by an average of 90%, which is 16% better than the improvements achieved by previous work. Efficiency of the frame memory recompression algorithm is improved with a 32×32 cache, resulting in a PSNR improvement of 0.371 dB. The H.264/AVC decoder with the three techniques is fabricated and implemented as an ASIC using 0.18 µm technology.

Bandwidth is an extremely valuable and scarce resource in multimedia networks. Therefore, efficient bandwidth management is necessary in order to provide high Quality of Service (QoS) to users. In this paper, a new QoS-aware bandwidth allocation algorithm is proposed for the efficient use of available bandwidth. By using the multi-objective optimization technique and Talmud allocation rule, the bandwidth is adaptively controlled to maximize network efficiency while ensuring QoS provisioning. In addition, we adopt the online feedback strategy to dynamically respond to current network conditions. With a simulation study, we demonstrate that the proposed algorithm can adaptively approximate an optimized solution under widely diverse traffic load intensities.

This paper describes a novel channel allocation and DBA (Dynamic Bandwidth Allocation) mechanism for ECDM-PON (Electric Code Division Multiplex -- Passive Optical Network) systems. In the current ECDM-PON systems, each ONU (Optical Network Unit) is limited to 2 or 3 CDM channels. This is because (fixed channel) CDM transmitters are expensive, and tunable CDM transmitters even more expensive. With a small number of CDM channels, the bandwidth utilization ratio is restricted by channel blocking. Our proposed mechanisms can reduce the channel blocking ratio without increasing the number of CDM transmitters or using tunable CDM transmitters. To clarify the advantages of the proposed system performance, we have evaluated the channel non-blocking ratio (Rn) and wasted resource ratio (Rw) when some users request bandwidth more than 100%. Evaluation of the non-blocking ratio, Rn shows that the proposed mechanisms approach the performance of a system with tunable CDM transmitters when the number of ONUs with over 100% traffic load is small. We have also simulated throughput for uniform traffic. In addition to these evaluations, we implemented our proposed mechanism on an FPGA (Field Programming Gate Array) and evaluated the calculation speed to allocate timeslots on CDM channels and a timeline.

A cognitive radio will have to sense and discover the spectral environments where it would not cause primary radios to interfere. Because the primary radios have the right to use the frequency, the cognitive radios as the secondary radios must detect radio signals before use. However, the secondary radios also need identifying the primary and other secondary radios where the primary radios are vulnerable to interference. In this paper, a method of simultaneously identifying signals of primary and secondary radios is proposed. The proposed bandwidth differentiation assumes the primary and secondary radios use orthogonal frequency division multiplexing (OFDM), and the secondary radios use at the lower number of subcarriers than the primary radios. The false alarm and detection probabilities are analytically evaluated using the characteristic function method. Numerical evaluations are also conducted on the assumption the primary radio is digital terrestrial television broadcasting. Result showed the proposed method could achieve the false alarm probability of 0.1 and the detection probability of 0.9 where the primary and secondary radio powers were 2.5 dB and 3.6 dB higher than the noise power. In the evaluation, the reception signals were averaged over the successive 32 snapshots, and the both the primary and secondary radios used QPSK. The power ratios were 4.7 dB and 8.4 dB where both the primary and secondary radios used 64QAM.

This paper presents two circuits to measure pulse width distribution of single event transients (SETs). We first review requirements for SET measurement in accelerated neutron radiation test and point out problems of previous works, in terms of time resolution, time/area efficiency for obtaining large samples and certainty in absolute values of pulse width. We then devise two measurement circuits and a pulse generator circuit that satisfy all the requirements and attain sub-FO1-inverter-delay resolution, and propose a measurement procedure for assuring the absolute width values. Operation of one of the proposed circuits was confirmed by a radiation experiment of alpha particles with a fabricated test chip.

A planar inverted-E (PIE) antenna that can achieve a wide impedance bandwidth is proposed. The antenna is realized by inserting a branch capacitance between the feed line and the shorting pin of a conventional planar inverted-F antenna (PIFA). Such a modification significantly enhanced the impedance bandwidth while maintaining the antenna size. The proposed antenna possesses a very wide impedance bandwidth of 1250 MHz (1650-2900 MHz) at a voltage standing wave ratio (VSWR) <3. In addition, good radiation patterns were obtained at the desired frequency bands.

In this letter, a new design of a metamaterial-based microstrip antenna is presented using triangular slots embedded on the ground plane to enhance the impedance bandwidth. To improve the impedance bandwidth of the proposed antenna, two resonant mode frequencies are closely allocated using the slotted ground without changing the radiator element. The impedance bandwidth of VSWR < 2.5 is measured at 2.43 GHz (37.6%) centered on 6.46 GHz, from 5.24 GHz to 7.67 GHz in good agreements with the simulated results.

The hybrid antenna consisted of cylindrical dielectric resonator and rectangular slot was implemented. The hybrid antenna resonated at two different frequencies. The lower resonant frequency was associated with the rectangular slot while the higher resonant frequency was associated with the cylindrical dielectric resonator. Parametric investigation was carried out using simulation software. The proposed hybrid antenna had good agreement between the simulation and measurement results. A 24% bandwidth (return loss < 10 dB) of 2.30 GHz, and a 18% bandwidth (return loss < 10 dB) of 5.46 GHz was implemented successfully for application in ISM and UNII band.

The IEEE 802.11 MAC standard for wireless ad hoc networks adopts Binary Exponential Back-off (BEB) mechanism to resolve bandwidth contention between stations. BEB mechanism controls the bandwidth allocation for each station by choosing a back-off value from one to CW according to the uniform random distribution, where CW is the contention window size. However, in asymmetric multi-hop networks, some stations are disadvantaged in opportunity of access to the shared channel and may suffer severe throughput degradation when the traffic load is large. Then, the network performance is degraded in terms of throughput and fairness. In this paper, we propose a new cross-layer scheme aiming to solve the per-flow unfairness problem and achieve good throughput performance in IEEE 802.11 multi-hop ad hoc networks. Our cross-layer scheme collects useful information from the physical, MAC and link layers of own station. This information is used to determine the optimal Contention Window (CW) size for per-station fairness. We also use this information to adjust CW size for each flow in the station in order to achieve per-flow fairness. Performance of our cross-layer scheme is examined on various asymmetric multi-hop network topologies by using Network Simulator (NS-2).

In this paper, VLSI architecture of a joint parameter decoder is proposed to realize the calculation of motion vector (MV), intra prediction mode (IPM) and boundary strength (BS) for ultra high definition H.264/AVC applications. For this architecture, a 64-cycle-per-MB pipeline with simplified control modes is designed to increase system throughput and reduce hardware cost. Moreover, in order to save memory bandwidth, the data which includes the motion information for the co-located picture and the last decoded line, is pre-processed before being stored to DRAM. A partition based storage format is applied to condense the MB level data, while variable length coding based compression method is utilized to reduce the data size in each partition. Experimental results show our design is capable of real-time 38402160@60 fps decoding at less than 133 MHz, with 37.2 k logic gates. Meanwhile, by applying the proposed scheme, 85-98% bandwidth saving is achieved, compared with storing the original information for every 44 block to DRAM.

Virtual Private Network (VPN) is a cost effective method to provide integrated multimedia services. Usually heterogeneous multimedia data can be categorized into different types according to the required Quality of Service (QoS). Therefore, VPN should support the prioritization among different services. In order to support multiple types of services with different QoS requirements, efficient bandwidth management algorithms are important issues. In this paper, I employ the Kalai-Smorodinsky Bargaining Solution (KSBS) for the development of an adaptive bandwidth adjustment algorithm. In addition, to effectively manage the bandwidth in VPNs, the proposed control paradigm is realized in a dynamic online approach, which is practical for real network operations. The simulations show that the proposed scheme can significantly improve the system performances.

Broadband wireless access networks are promising technology for providing better end user services. For such networks, designing a scheduling algorithm that fairly allocates the available bandwidth to the end users and maximizes the overall network throughput is a challenging task. In this paper, we develop a centralized fair scheduling algorithm for IEEE 802.16 mesh networks that exploits the spatio-temporal bandwidth reuse to further enhance the network throughput. The proposed mechanism reduces the length of a transmission round by increasing the number of non-contending links that can be scheduled simultaneously. We also propose a greedy algorithm that runs in polynomial time. Performance of the proposed algorithms is evaluated by extensive simulations. Results show that our algorithms achieve higher throughput than that of the existing ones and reduce the computational complexity.

This paper presents an error compensation method for fixed-width group canonic signed digit (GCSD) multipliers that receive a W-bit input and generate a W-bit product. To efficiently compensate for the truncation error, the encoded signals from the GCSD multiplier are used for the generation of the error compensation bias. By Synopsys simulations, it is shown that the proposed method leads to up to 84% reduction in power consumption and up to 78% reduction in area compared with the fixed-width modified Booth multipliers.

Recently, network coding has been applied to the loss recovery of reliable multicast in wireless networks, where multiple lost packets are XOR-ed together as one packet and forwarded via single retransmission, resulting in a significant reduction of bandwidth consumption. In this paper, we first prove that maximizing the number of lost packets for XOR-ing, which is the key part of the available network coding-based reliable multicast schemes, is actually a complex NP-complete problem. To address this limitation, we then propose an efficient heuristic algorithm for finding an approximately optimal solution of this optimization problem. Furthermore, we show that the packet coding principle of maximizing the number of lost packets for XOR-ing sometimes cannot fully exploit the potential coding opportunities, and we then further propose new heuristic-based schemes with a new coding principle. Simulation results demonstrate that the heuristic-based schemes have very low computational complexity and can achieve almost the same transmission efficiency as the current coding-based high-complexity schemes. Furthermore, the heuristic-based schemes with the new coding principle not only have very low complexity, but also slightly outperform the current high-complexity ones.

A virtualized optical network (VON) is proposed as a key to implementing increased agility and flexibility into the future Internet and applications by providing any-to-any connectivity with the appropriate optical bandwidth at the appropriate time. The VON is enabled by introducing optical transparentization and optical fine granular grooming based on optical orthogonal frequency division multiplexing.

Motion compensation is widely used in many video coding standards. Due to its bandwidth requirement and complexity, motion compensation is one of the most challenging parts in the design of high definition video decoder. In this paper, we propose a high performance and low bandwidth motion compensation design, which supports H.264/AVC, MPEG-1/2 and Chinese AVS standards. We introduce a 2-Dimensional cache that can greatly reduce the external bandwidth requirement. Similarities among the 3 standards are also explored to reduce hardware cost. We also propose a block-pipelining strategy to conceal the long latency of external memory access. Experimental results show that our motion compensation design can reduce the bandwidth by 74% in average and it can real-time decode 1920x1088@30 fps video stream at 80 MHz.

Recent research on overlay networks has revealed that user-perceived network performance could be improved by an overlay routing mechanism. The effectiveness of overlay routing is mainly a result of the policy mismatch between the overlay routing and the underlay IP routing operated by ISPs. However, this policy mismatch causes a "free-riding" traffic problem, which may become harmful to the cost structure of Internet Service Providers. In the present paper, we define the free-riding problem in the overlay routing and evaluate the degree of free-riding traffic to reveal the effect of the problem on ISPs. We introduce a numerical metric to evaluate the degree of the free-riding problem and confirm that most multihop overlay paths that have better performance than the direct path brings the free-riding problem. We also discuss the guidelines for selecting paths that are more effective than the direct path and that mitigate the free-riding problem.

Currently deployed mobile networks including High Speed Downlink Packet Access (HSDPA) offer only best-effort Quality of Service (QoS). In wireless best effort networks, the bandwidth variation is a critical problem, especially, for mobile devices with small buffers. This is because the bandwidth variation leads to packet losses caused by buffer overflow as well as picture freezing due to high transmission delay or buffer underflow. In this paper, in order to provide seamless video streaming over HSDPA, we propose an efficient real-time video streaming method that consists of the available bandwidth (AB) estimation for the HSDPA network and the transmission rate control to prevent buffer overflows/underflows. In the proposed method, the client estimates the AB and the estimated AB is fed back to the server through real-time transport control protocol (RTCP) packets. Then, the server adaptively adjusts the transmission rate according to the estimated AB and the buffer state obtained from the RTCP feedback information. Experimental results show that the proposed method achieves seamless video streaming over the HSDPA network providing higher video quality and lower transmission delay.

In recent years, there have been many studies on integrating a number of heterogeneous wireless networks into one network by establishing standards like IEEE 802.16. For this purpose, the base station (BS) should allocate the appropriate bandwidth to each connection with a network scheduler. In wireless networks, the signal to noise ratio (SNR) changes with time due to many factors such as fading. Hence, we estimate the SNR based on the error rate reflecting wireless network condition. Using the estimated SNR, we propose a new time slot allocation algorithm so that the proposed algorithm guarantees the delay requirement and full link utilization.

In this paper, we propose a bandwidth-scalable stereo audio coding method based on a layered structure. The proposed stereo coding method encodes super-wideband (SWB) stereo signals and is able to decode either wideband (WB) stereo signals or SWB stereo signals, depending on the network congestion. The performance of the proposed stereo coding method is then compared with that of a conventional stereo coding method that separately decodes WB or SWB stereo signals, in terms of subjective quality, algorithmic delay, and computational complexity. Experimental results show that when stereo audio signals sampled at a rate of 32 kHz are compressed to 64 kbit/s, the proposed method provides significantly better audio quality with a 64-sample shorter algorithmic delay, and comparable computational complexity.