The accuracy of channel estimation significantly affects the performance of coherent rake receiver in DS-CDMA systems. It is desirable for improved channel estimation to employ a channel estimation filter (CEF) whose bandwidth is adjustable to the channel condition. In this paper, we consider the use of moving average (MA) FIR filters as the CEF since it is simple to implement and can provide relatively good receiver performance. First, we optimize the tap size of the MA FIR CEF so as to minimize the mean squared error of the estimated channel impulse response. For practical applications, we propose a low-complexity adaptive channel estimator (ACE), where the tap size of the MA FIR CEF is adjusted based on the estimated channel condition by exploiting the correlation characteristics of the received pilot signal. Numerical results show that the use of the proposed ACE can provide the receiver performance comparable to that of Wiener CEF without exact a priori information on the operating condition.

TCP (Transmission Control Protocol) is the prevalent reliable transport protocol used for many kinds of the popular Internet services like web browsing and e-mail. Because these services are the dominant applications on the Internet, TCP can be said to control the large majority of today's Internet traffic and we can say the enhancement of TCP performance means the improvement of the most Internet performance. This paper compares various ACK generation TCP strategies that play very important roles in TCP operation and proposes advanced delayed ACK algorithms to mitigate the negative effect of delayed ACK algorithm. Even if delayed ACK has the bright side to save network resources and power consumption that are the very important factors especially for mobile and wireless environment, it has also dark side not to take full advantage of network bandwidth in Slow Start phase. It is reported that delayed ACK timeout of the first and last segment causes severe degradation problem in TCP throughput performance. We suggest new algorithms to solve delayed ACK penalty and to guarantee fairness with other traffics in Slow Start. In wired-cum-wireless network environments,'BADA (Base station Aided Delayed ACK)' with SNDA (Sender Notification Delayed ACK), the combination of our proposed algorithm shows the best performance in terms of throughput as well as in terms of network resource sharing. Our simulation shows significantly improved performance in TCP throughput.

This paper describes a design of a compact active loop filter for Phase-Locked-Loop (PLL) with adaptive biasing technique. Using the new loop filter, the PLL can automatically adjust the loop bandwidth and damping factor to the frequency of the reference clock. Moreover, the new LPF can decrease the capacitance value to 1/10-1/20 of conventional one. A test chip was fabricated in 0.15 µm-CMOS process. The total chip area of the PLL is reduced to 1/2 of the previous one. The jitter performance is almost equal to conventionally biased PLL.

Modern spaceborne precision radar altimeters transmit radio signals of a spectrum bandwidth up to 300 MHz, but the bandwidth should be still increased for precise estimation of the roughness of the sea surface. In this research, the influence of the ionosphere on wideband radar signals is investigated and then it is shown that the signals are strongly influenced by the dispersive distortions in the atmosphere of the Earth even in Ku-band. Finally, the allowable bandwidth of a space borne precision radar altimeter signals is estimated, at which we could ignore the presence of these distortions.

In this paper, we perform some experiments to show that the quantization noise caused by low-bit-rate speech coding can be characterized as a white noise process. Then, the signal-to-quantization noise ratio of the decoded speech for a given bit-rate is estimated by observing the perceptual speech quality equivalent to the artificially generated noisy speech obtained by adding a white Gaussian noise source. This information is incorporated into the parameter tuning of a noise-robust compensation algorithm for speech recognition so that the compensation algorithm can be performed better under a range of the estimated SNRs. Finally, we apply the compensation algorithm to a connected digit string recognition system that utilizes speech signals decoded by the GSM adaptive multi-rate (AMR) speech coder. It is shown that the noise-robust compensation algorithm reduces word error rates by 15% or more at low bit-rate modes of the AMR speech coder.

In this letter, adaptive modulation and coding (AMC) is combined with multiple input multiple output (MIMO) multiplexing to improve the throughput performance of AMC. In addition, a system that adopts selection transmit diversity (STD) in the AMC-MIMO multiplexing system, AMC-STD-MIMO multiplexing system, is proposed. STD in our simulation selects 2 transmission antennas from 4 antennas and AMC-MIMO multiplexing process operates with the selected antennas. The computer simulation is performed in a flat Rayleigh fading channel. The results show that the proposed AMC-STD-MIMO multiplexing system achieves the SNR gain of 4.0 dB, compared to the AMC-MIMO multiplexing system at 3 Mbps throughput.

Internet traffic engineering is much important for Internet Service Providers (ISPs) today, since it can be used to fully utilize already deployed network resources. For ISPs, the requirements for traffic engineering should be simple, easy to configure, cost-effective and efficient. Based on these considerations, we propose an algorithm called Web First Adaptive Traffic Engineering (WFATE). Since World Wide Web (WWW) services dominate most of the total Internet traffic and WWW flows are not long-lived, we only apply load balancing to WWW traffic in the algorithm. It can be shown that the number of coexistent WWW flows at an ingress node is almost certainly below a bound, and thus a forward-per-flow mechanism without keeping track of the state of each flow is feasible. This mechanism can balance traffic load at fine granularity and therefore get better performance. Through simulations and performance comparison, it is shown that WFATE is quite efficient, which can improve the network throughput averagely by 26% under the "dense source" traffic pattern and 9% under the "sparse source" traffic pattern.

Radars utilizing ultra-wide-band (UWB) pulses are attractive as an environment measurement method for various applications including household robots. Suitable filtering is essential for accurate ranging, which requires an accurate waveform estimation. This paper presents a high-resolution algorithm of estimating target location and scattered waveforms, whose accuracies are interdependent. The technique relies on iterative improvements of estimated waveforms. Description of the algorithm is followed by statistical simulation examples. The performance of the algorithm is contrasted with conventional ones and statistical bounds. Results indicate that our proposed algorithm has a remarkable performance, which is close to the theoretical limit. Next, we clarify the problem of applying HCT to multiple targets. HCT for multiple targets can not be used as an estimated waveform because of interference waves from other targets. We propose an interference suppression algorithm based on a neural network, and show an application example of the algorithm.

We propose a new data link protocol with an adaptive frame length control scheme for satellite networks. The wireless communication channel in satellite networks is subject to errors that occur with time variance. The frame length of the data link layer is another important factor that affects throughput performance in dynamic channel environments. If the frame length could be chosen adaptively in response to changes in the dynamically varying satellite channel, maximum throughput could be achieved under both noisy and non-noisy error conditions. So, we propose a frame length control scheme that acts adaptively to counter errors that occur with time variance. We model the satellite channel as a two-state Markov block interference (BI) model. The estimation of the channel error status is based on the short-term bit error rate and the duty cycle of noise bursts. Numerical and computer simulation results show that the proposed scheme can achieve high throughput for both dense and diffuse burst noise channels.

In this letter, we propose an adaptive turbo coded modulation scheme for orthogonal frequency division multiplexing (OFDM), and three power allocation algorithms with transmit power variable over subcarriers, variable over subbands and constant over subcarriers. Our object is to improve the overall throughput under target bit-error-rate (BER). Simulation results show that our fixed power adaptation scheme exhibits a more than 7 dB signal-noise-ratio (SNR) gain relative to non-adaptive turbo coded modulation, and an about 2 dB additional gain can be achieved with our variable power algorithm employed. We also discuss the effect of the number of subbands on throughput performance in our adaptive scheme.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

Environment measurement is an important issue for various applications including household robots. Pulse radars are promising candidates in a near future. Estimating target shapes using waveform data, which we obtain by scanning an omni-directional antenna, is known as one of ill-posed inverse problems. Parametric methods such as Model-fitting method have problems concerning calculation time and stability. We propose a non-parametric algorithm for high-resolution estimation of target shapes in order to solve the problems of parametric algorithms.

In this paper, an adaptive transmission scheme considering the stay time in a spot mobile access system is proposed. The proposed adaptive transmission scheme selects the modulation format according to the user's stay time in the spot communication zone and the types of data requested by each user. In the proposed system, when the stay time of a user is short, high-speed modulation is selected for this user. When the stay time of a user is long, a more reliable modulation format is selected. The computer simulation results show that the proposed transmission scheme without any channel estimation can achieve the same or better performance than when using the modulation format fixedly when the carrier-to-noise ratio changes rapidly.

This paper proposes the constraint availability and bandwidth shortest path (CABSP) selection algorithm and the extension of the adaptive modulation scheme to the CABSP (CABSP-AM) selection algorithm in the millimeter-wave (MMW) broadband entrance network for wireless heterogeneous systems. The CABSP algorithm considers the availability denoted as the quality of the MMW which is severely affected by the rainfall. The CABSP-AM algorithm, moreover, is proposed to further make more efficient use of bandwidth resources by considering the QoS requirements of each class of service in multimedia communication. As the results, the CABSP algorithm yields higher throughput performance than the conventional constraint shortest path (CSP) selection algorithm, but lower than the CABSP-AM algorithm. The spectrum efficiency improvements of the CABSP-AM over the CABSP are about 1.36 and 1.48 fold in case of error sensitive and non-sensitive classes respectively.

In this paper, a simple blind algorithm for a beamforming antenna is proposed. This algorithm exploits the property of cyclostationary signals whose cyclic autocorrelation function depends on delay as well as frequency. The cost function is the mean square error between the delay product of the beamformer output and a complex exponential. Exploiting the delay greatly reduces the possibility of capturing undesired signals. Through analysis of the minima of the non-quadratic cost function, conditions to extract a single signal are derived. Application of this algorithm to code-division multiple-access systems is considered, and it is shown through simulation that the desired signal can be extracted by appropriately choosing the delay as well as the frequency.

This paper describes an adaptive fingerprint-sensing scheme for a user authentication system with a fingerprint sensor LSI to obtain high-quality fingerprint images suitable for identification. The scheme is based on novel evaluation indexes of fingerprint-image quality and adjustable analog-to-digital (A/D) conversion. The scheme adjusts dynamically an A/D conversion range of the fingerprint sensor LSI while evaluating the image quality during real-time fingerprint-sensing operation. The evaluation indexes pertain to the contrast and the ridgelines of a fingerprint image. The A/D conversion range is adjusted by changing quantization resolution and offset. We developed a fingerprint sensor LSI and a user authentication system to evaluate the adaptive fingerprint-sensing scheme. The scheme obtained a fingerprint image suitable for identification and the system achieved an accurate identification rate with 0.36% of the false rejection rate (FRR) at 0.075% of the false acceptance rate (FAR). This confirms that the scheme is very effective in achieving accurate identification.

We propose adaptive variable-rate constant-power scheme for ad hoc wireless networks, employing the modification of Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) protocol. Potential improvements in throughput and back-off probability are presented for different system parameters.

A novel thresholding algorithm for change detection in video sequences is proposed. The method is based on image differencing and the intensity distribution of a difference image. With a difference image between two consecutive images, we prepare a new image model for the distribution of stationary pixels. The distribution of moving pixels is then separated by extracting the distribution of stationary pixels from the overall distribution of the difference image. Pixels that exhibit a significant change in intensity are classified using a likelihood criterion. The proposed algorithm is tested on the standard MPEG sequences and verified to have reliable performance.

Multi-rate capabilities are supported by the physical layers of most 802.11 devices. To enhance the network throughput of MANETs, transfer rate adaptation schemes at MAC layer should employ the multi-rate capability at physical and the information of previous transmissions provided by MAC and physical layers. In this paper, we propose a transfer rate adaptation scheme plus back-to-back frame transmissions, and fragmentation at MAC layer, named TRAF. TRAF adopts a bi-direction-based approach with an extended option to select an appropriate rate for frame transmission under fast changing channel conditions. Consecutive back-to-back frame transmissions to fully utilize good channel quality during a coherent time interval and fragmentation algorithm to maintain high throughput under worse channel conditions are recommended in TRAF. Extensive simulation is experimented to evaluate the performance of TRAF. Regarding simulation results, frame delivery ratio, network throughput, and fairness of TRAF are significantly improved by comparing to that of fix rate, ARF, RBAR, OAR, and AAR protocols.

In this paper, we study an effective video-on-demand traffic control algorithm using the metadata over the network supporting bandwidth-renegotiations. The proposed algorithm includes bandwidth smoothing, bandwidth-burstiness estimation and rate adaptation algorithms. The proposed video-on-demand server has not only video database but also metadata database that includes coding information of the compressed video in video database and the traffic burst characteristics with respect to control parameters of the bandwidth smoothing algorithm. Thus, we can predict the traffic properties accurately with a low computational complexity by using the stored metadata, and then determine the efficient bandwidth renegotiating variables such as the renegotiating instants and the required bandwidth in terms of network utilization and video-on-demand service quality. In addition, we present a rate adaptation algorithm that pursues an effective trade-off between spatial and temporal qualities of the decoded video to improve the perceptual video quality when the bandwidth request is rejected.