This paper presents an ultra-wideband (UWB) bandpass filter using a combination of broadside-coupled structure and lumped-capacitor-loaded shunt stub resonator. The broadside-coupled microstrip-to-coplanar waveguide structure provides an ultra-wide bandpass filtering operation and keeps a good stopband at lower frequencies from DC at the same time. The lumped-capacitor-loaded shunt stub resonator creates two transmission zeros (attenuation poles which can be located at the outsides of the two bandedges of the UWB bandpass filter to improve the out-band performance by selecting a suitable combination of the length of the shunt stubs and the capacitance of the loaded chip capacitors. The filter was designed based on electromagnetic simulation for broadside-coupled structure, microwave circuit simulation and experiments for determining the transmission zeros. The filter was fabricated on a one-layer dielectric substrate. The measured results demonstrated that the developed UWB bandpass filter has good performance: low insertion loss about 0.46 dB and low group delay about 0.26 ns at the center of the passband and very flat over the whole passband, and less than -10 dB reflection over the passband. The implemented transmission zeros, particularly at the low frequency end, dramatically improved the out-band performance, leading the filter satisfy the FCC's spectrum mask not only for indoor but also for outdoor applications. These poles improved also the skirt performance at both bandedges of the filter. A lowpass filter has been also introduced and integrated with the proposed bandpass filter to have a further improvement of the out-band performance at the high frequency end. The filters integrated with lowpass section exhibit excellent filter performance: almost satisfying the FCC's spectrum mask from DC to 18 GHz. The developed UWB bandpass filter has a compact size of 4 cm1.5 cm, or 4.8 cm1.5 cm with lowpass section implemented.

In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pair (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods by investigating the distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. Next, the proposed techniques are applied to the predictive vector quantizer (PVQ) used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064 dB and the percentage of outliers compared with the PVQ without any compensation, resulting in transparent quality of spectral quantization. Finally, the comparison of speech quality using the perceptual evaluation of speech quality (PESQ) measure is performed and it is shown that the IS-641 speech coder employing the proposed techniques has better decoded speech quality than the standard IS-641 speech coder.

Based on the mutuality between arrival moments of uplink and downlink messages, this paper proposes a scheme that assigns different time-out thresholds for mobile terminal sleep mode operation according to the direction of the message just processed. Simulation results prove that, this approach can increase the power saving factor of a mobile terminal without degrading QoS.

Due to the discontinuity of packet based traffic, the user terminals in next generation mobile telecommunications systems will be equipped with sleep mode operation functions for power saving purpose. The sleep mode parameters should be appropriately configured so that power consumption can be sufficiently decreased while packet queue length and packet delay are restricted within a demanded level. This paper proposes an adaptive sleep mode parameter configuration scheme which is able to jointly optimize the inactivity timer and sleep period in response to the variation of user traffic arrival pattern. The optimization target of this scheme is to minimize mobile terminal power consumption while ensuring that the mean downlink packet queue length do not exceed a certain threshold. Results of computer simulations prove that, the presented approach perfectly manages packet queue length restriction, packet delay control and power saving in a wide range of user packet inter-arrival rates both in single- and dual-service scenarios.

Orthogonal frequency division multiplexing (OFDM) systems for mobile applications suffer from inter-carrier-interference (ICI) due to frequency offset and to time-variation of the channels and from high peak-to-average-power ratio (PAPR). In this paper, we revisit symmetric cancellation coding (SCC) proposed by Sathananthan et al. and compare the effectiveness of SCC with a fixed subtraction combining and the well-known polynomial cancellation coding (PCC) over Rayleigh fading channels with Doppler spread in terms of the signal-to-interference plus noise power ratio (SINR) and bit-error-rate (BER). We also compare SCC with subtraction combining and SCC of Sathananthan et al. with maximum ratio combining (MRC). Our results show that SCC-OFDM with subtraction combining gives higher SINR than PCC-OFDM over the flat Rayleigh fading channel and that this superiority is not maintained under multi-path induced frequency-selective fading unless diversity combining is used. A simulation result shows, however, that SCC-OFDM with subtraction combining may perform better than PCC-OFDM for a certain range of Doppler spread when differential modulation is employed. Finally, we also demonstrate that the SCC-OFDM signal has less PAPR compared to the normal OFDM and PCC-OFDM and hence may be more practical.

The performance of a real-time networked application can be drastically affected by delays in packets traversing the network. Some real-time applications impose limits for acceptable network delay, and so a packet which is delayed longer than the limit before arriving at its destination is worthless to the flow to which the packet belongs. Not only that, but the rejected packet is also damaging to the quality of other flows in the network, because it may increase the queuing delay for other packets. Therefore, this paper proposes an adaptive scheme using two mechanisms, in which packets experiencing too great a delay are discarded at intermediate nodes based on the delay limit for the application and the delay experienced by each packet. This earlier discarding of packets is expected to improve the overall delay performance of real-time flows competing for network resources when the network is congested. An extensive simulation is conducted, and the results show that the scheme has great potential in improving the delay performance of real-time traffic in both homogeneous and heterogeneous environments in terms of traffic volume and application delay requirements.

One of the interesting submicron MOS FET characteristics is the effect of carrier velocity saturation (CVS) on the drain current. In the CVS region, the transconductance becomes constant independent both of the gate and the drain voltage. In this paper, RF MOS amplifier design technique using the CVS region has been proposed. By setting the FET gate bias to the power supply voltage Vdd, stable operation against Vdd variations can be achieved with a simple circuit configuration. By using this, a 5 GHz amplifier has been designed and fabricated by using 0.18-µm CMOS process technology. The chip has been operated with a gain variation less than 1 dB having a peak gain of 13.5 dB from 1.2 to 2.9 V Vdd.

Direction of arrival estimation of coherent multipath waves by using superresolution technique often requires decorrelation preprocessings. Spatial smoothing preprocessings are the most popular schemes as the techniques. In mobile environment, position change of the target/transmitter often brings us decorrelation effect. In addition, multiple signals transmitted by an antenna array, such as a MIMO transmitter, can also cause the same effect. These effects can be categorized as the spatial smoothing preprocessing at the transmitter. In this paper, we analyze the spatial smoothing effect at the transmitter in the presence of multipath coherent waves. Theoretical and simulation results show that the spatial smoothing at the transmitter has a good feature in comparison with the conventional SSP at the receiving array. We also show that better decorrelation performance can be obtained when the SSPs at the transmitter and receiving array are applied simultaneously.

China has experienced fast growth in mobile communications. Now, China is the world largest mobile communication country with about 500 million users. Wide applications of mobile communications are giving strong pull to the research and development on the broadband wireless communication technology to meet the fast growing demand for high speed access into the information infrastructure. This makes the R&D on wireless technology play great role in the Chinese High-Tech program. This paper will review the key project--FuTURE (Future Technology for Universal Radio Environment)--development of the 863 program, which represents the Chinese efforts towards IMT-advance. Taking some works done in the Tsinghua National Laboratory for Information Science and Technology as examples, the paper will show what has been made in China on the broadband wireless technology, including the trial network in Shanghai.

This paper proposes a new binary motion estimation algorithm that improves the motion vector accuracy by using a hybrid distortion measure. Unlike conventional binary motion estimation algorithms, the proposed algorithm considers the sum of absolute difference (SAD) as well as the sum of bit-wise difference (SBD) as a block-matching criterion. In order to reduce the computational complexity and remove additional memory accesses, a new scheme is used for SAD calculation. This scheme uses 8-bit data of the lowest layer already moved into the local buffer to calculate the SAD of other higher binary layer. Experimental results show that the proposed algorithm finds more accurate motion vectors and removes the blockishness of the reconstructed video effectively. We applied this algorithm to existing video encoder and obtained noticeable visual quality enhancement.

This letter proposes a sliding window method with iterative tuning for channel estimation of UWB signals. The iterative tuning scheme, which is based on multiple iterations of least mean square (LMS) algorithm, is utilized for modifying the output of the conventional sliding window channel estimator. By using this, the proposed method is more flexible due to the tradeoff between the processing time and accuracy, which makes it more suitable for practical UWB wireless communications. Simulations are also provided for demonstrating the validation of the proposed method.

In the main part of this paper, we present a systematic discussion for the optimum interpolation approximation in a shift-invariant wavelet and/or scaling subspace. In this paper, we suppose that signals are expressed as linear combinations of a large number of base functions having unknown coefficients. Under this assumption, we consider a problem of approximating these linear combinations of higher degree by using a smaller number of sample values. Hence, error of approximation happens in most cases. The presented approximation minimizes various worst-case measures of approximation error at the same time among all the linear and the nonlinear approximations under the same conditions. The presented approximation is quite flexible in choosing the sampling interval. The presented approximation uses a finite number of sample values and satisfies two conditions for the optimum approximation presented in this paper. The optimum approximation presented in this paper uses sample values of signal directly. Hence, the presented result is independent from the so-called initial problem in wavelet theory.

In this letter, we design an optimal superimposed training scheme for orthogonal frequency division multiplexing (OFDM) systems. A linear least square (LS) channel estimator is developed, and optimal pilot symbols are proposed with respect to the channel estimate's mean square error (MSE). The optimal power allocation between training and data is derived by maximizing the averaged channel capacity lower bound. Simulation results validate our optimum design.

As a fundamental building block of multirate systems, the downsampler, also known as the decimator, is a periodically time-varying linear system. An eigensignal of the downsampler is defined to be an input signal which appears at the output unaltered or scaled by a non-zero coefficient. In this paper, the eigensignals are studied and characterized in the time and z domains. The time-domain characterization is carried out using number theoretic principles, while the one-sided z-transform and Lambert-form series are used for the transform-domain characterization. Examples of non-trivial eigensignals are provided. These include the special classes of multiplicative and completely multiplicative eigensignals. Moreover, the locus of poles of eigensignals with rational z transforms are identified.

In recent years, wireless communications systems such as wireless LAN, Bluetooth, etc. are being rapidly adopted. As the antennas used in wireless communications systems are usually installed in small mobile devices, it is demanded that the volume should be small. In our research, we focus our attention on flexible printed circuits (FPCs) to meet the miniaturization demand. In this paper, we introduce a planar inverted F antenna (PIFA) suitable for IEEE802.11b/g and Bluetooth. The antenna is made of FPC. We measured the radiation pattern of the antenna when the antenna is successively curved and folded, and it is clear that its radiation performance does not vary much when the antenna is deformed. We analyzed the antenna by using the moment method.

The error resilient entropy coding (EREC) provides efficient resynchronization method to the coded bitstream, which might be corrupted by transmission errors. The technique has been given more prominence, nowadays, because it achieves fast resynchronization without sizable overhead, and thereby provides graceful quality degradation according to the network conditions. This paper presents a novel framework to analyze the performance of EREC in terms of the error probability in decoding a basic resynchronization unit (RU) for various error prone networks. In order to show the feasibility of the proposed framework, this paper also proposes a novel EREC algorithm based on the slightly modified H.263 bitstream syntax. The proposed scheme minimizes the effect of errors on low frequency DCT coefficients and incorporates near optimal channel-matched searching pattern (SP), which guarantees the best possible quality of reproduced video. Given the number of bits generated for each RU, the near optimal SP is produced by the proposed iterative deterministic partial SP update method, which reduces the complexity of finding optimal solution, O((N-1)!), to O(m·N2). The proposed EREC algorithm significantly improves the decoded video quality, especially when the bit error rate is in the rage of 10-3-10-4. Up to 5 dB enhancement of the PSNR value was observed in a single video frame.

Collaboration of growing self-organizing maps (GSOM) and adaptive resonance theory maps (ART) is considered through traveling sales-person problems (TSP).The ART is used to parallelize the GSOM: it divides the input space of city positions into subspaces automatically. One GSOM is allocated to each subspace and grows following the input data. After all the GSOMs grow sufficiently they are connected and we obtain a tour. Basic experimental results suggest that we can find semi-optimal solution much faster than serial methods.

Drivability-improved MOSFETs were successfully fabricated by using nano-grating silicon wafers. There was almost no additional process change in device fabrication when the height of the gratings was less than the conventional macroscopic wafer surface roughness. The MOSFETs with the grating height of 35 nm showed 21% improvement in current drivability compared to the conventional one with the same device occupancy area. And the roll-off characteristic of threshold voltage of nano-grating device held the line of conventional one in despite of the 3-D channel structure. The technology provides great advantages for drivability improvement without paying much tradeoff of process cost. This proposal will be useful to CMOS-LSIs with high performance in general.

A new type of sound source segregation method using robot-mounted microphones, which are free from strict head related transfer function (HRTF) estimation, has been proposed and successfully applied to three simultaneous speech recognition systems. The proposed segregation method is executed with sound intensity differences that are due to the particular arrangement of the four directivity microphones and the existence of a robot head acting as a sound barrier. The proposed method consists of three-layered signal processing: two-line SAFIA (binary masking based on the narrow band sound intensity comparison), two-line spectral subtraction and their integration. We performed 20 K vocabulary continuous speech recognition test in the presence of three speakers' simultaneous talk, and achieved more than 70% word error reduction compared with the case without any segregation processing.

DHT based p2p systems appear to provide scalable storage services with idle resource from many unreliable clients. If a DHT is used in storage intensive applications where data loss must be minimized, quick replication is especially important to replace lost redundancy on other nodes in reaction to failures. To achieve this easily, a simple replication method directly uses a consistent set, such as a leaf set and a successor list. However, this set is tightly coupled to the current state of nodes and the traffic needed to support this replication can be high and bursty under churn. This paper explores efficient replication methods that only glimpse a consistent set to select a new replica. Replicas are loosely coupled to a consistent set and we can eliminate the compulsory replication under churn. Because of a complication of the new replication methods, the careful data management is needed under churn for the correct and efficient data lookup. Results from a simulation study suggest that our methods can reduce network traffic enormously for high data durability.