A high-speed triangular-modulated spread-spectrum clock generator using a fractional phase-locked loop is presented. The fractional division is implemented by a nested fractional topology, which is constructed from a dual-modulus divide-by-(N-1/16)/N divider to divide the VCO outputs as a first division period and a fractional control circuit to establish a second division period to cause the overall fractional division. The dual-modulus divider introduces a delay-locked-loop network to achieve phase compensation. Operating at the frequency of 3.2 GHz, the measured peak power reduction is around 16 dB for a deviation of 0.37% and a frequency modulation of 33 kHz. The circuit occupies 1.41.4 mm2 in a 0.18-µm CMOS process and consumes 52 mW.

This paper extends previous analytical approaches for the study of CDMA systems to the relevant case of multipath environments where users can operate at different bit rates. This scenario is of interest for the Wideband CDMA strategy employed in UMTS, and the model permits the performance comparison of classic and more innovative spreading signals. The method is based on the characteristic function approach, that allows to model accurately the various kinds of interferences. Some numerical examples are given with reference to the ITU-R M.1225 Recommendations, but the analysis could be extended to different channel descriptions.

A method for extracting a sound signal from a particular area that is surrounded by multiple ambient noise sources is proposed. This method performs several fixed beamformings on a pair of small microphone arrays separated from each other to estimate the signal and noise power spectra. Noise suppression is achieved by applying spectrum emphasis to the output of fixed beamforming in the frequency domain, which is derived from the estimated power spectra. In experiments performed in a room with reverberation, this method succeeded in suppressing the ambient noise, giving an SNR improvement of more than 10 dB, which is better than the performance of the conventional fixed and adaptive beamforming methods using a large-aperture microphone array. We also confirmed that this method keeps its performance even if the noise source location changes continuously or abruptly.

In this paper, sampling rate selection diversity (SRSD) scheme for Direct-Sequence/Spread-Spectrum (DS/SS) is proposed. In DS/SS communication systems, oversampling may be employed to increase the signal-to-noise ratio (SNR). However, oversampling enlarges the power consumption because signal processing of the receiver has to be carried out at a higher clock rate. Higher sampling rate does not always maximize the SNR. In the proposed SRSD scheme, the power consumption can be reduced by selecting the optimum sampling rate depending on the characteristics of the channel. The proposed SRSD scheme can also reduce the BER more than the conventional oversampling scheme under certain channel conditions.

In this paper, we propose and demonstrate a novel type of PCF that has two cladding layers with Ge rods at the center core. We numerically show that it is possible to design a single mode PCF with large effective area greater than 200 µm2 over the whole wavelength above 1.2 µm. The proposed large mode area PCF (LMA-PCF) exhibits a high negative dispersion coefficient from -186 to -158 [ps/(nm-km)] in all wavelengths ranging from 1.2 µm to 1.8 µm. Effective single mode operation of LMA-PCF is confimed for the entire band of interest.

Cognitive Radios (CR) can recognize the communication environment and switch its communication scheme to more efficiently and flexibly utilize the radio spectrum. The performance of ultra wideband (UWB) degrades if interference is not suppressed properly. We propose here a series of adaptive chirp waveforms in UWB systems. By designing waveform shaping of both linear chirp and non-linear cases, we avoid the estimated spectrum of the on-going applications without the necessity of notch filters, and thus reduce the system complexity. We evaluate system performance of the proposed scheme by simulations and verify that the proposed scheme is a candidate for cognitive UWB systems.

In a Cognitive Radio system, it is essential to recognize and avoid sources of interference signals. This paper describes a study on a location sensing scheme for interference signals, which utilizes multi-beam phased array antenna for cognitive wireless networks. This paper also elucidates its estimation accuracy of the interference location for the radio communication link using an OFDM signal such as WiMAX. Furthermore, we use the frequency spectrum of the received OFDM interference signal, to create a method that can estimate the propagation status. This spectrum can be monitored by using a software defined radio receiver.

We propose a novel asynchronous direct-sequence code-division multiple access (DS-CDMA) using feedback-controlled spreading sequences (FCSSs) (FCSS/DS-CDMA). At the receiver of FCSS/DS-CDMA, the code-orthogonalizing filter (COF) produces a spreading sequence, and the receiver returns the spreading sequence to the transmitter. Then the transmitter uses the spreading sequence as its updated version. The performance of FCSS/DS-CDMA is evaluated over time-dispersive channels. The results indicate that FCSS/DS-CDMA greatly suppresses both the intersymbol interference (ISI) and multiple access interference (MAI) over time-invariant channels. FCSS/DS-CDMA is applicable to the decentralized multiple access.

In this paper, a fiber with two inhomogeneous sector holes around the core is proposed, and propagation characteristics of polarization maintaining region and single-polarization region are numerically analyzed by circular Fourier expansion method. In each case of the single-polarization region and the polarization maintaining region, a fiber is designed so as to satisfy the zero total dispersion at wavelength of 1.55 µm. Then, the single-polarization bandwidth for the single-polarization region and the modal birefringence for the polarization maintaining region are examined as the specific characteristics in each region. In addition, the power concentrating into the core region and distributions of Poynting vector is also discussed.

We have derived the novel extended UTD (Uniform Geometrical Theory of Diffraction) solution and the novel modified UTD solution for the back scattering of an incident whispering gallery (WG) mode on the edge of a cylindrically curved conducting sheet. By comparing with the reference solution obtained from the integral representation of the scattered field by integrating numerically along the integration path, we have confirmed the validity and the utility of the novel asymptotic solutions proposed in the present study. It is shown that the extended UTD solution can be connected smoothly to the modified UTD solution on the geometrical boundary separating the edge-diffracted ray and the surface-diffracted ray.

In this paper, an architecture of MIMO mesh network which avoids co-channel interference and supplies link multiplexing simultaneously, namely MIMO spatial spectrum sharing, is proposed. As a MIMO transmission scheme, linear (such as zero-forcing) and nonlinear (such as dirty paper coding and successive interference cancellation) MIMO algorithm are developed for the proposed mesh network. It is found from numerical analysis that the proposed MIMO mesh network achieves significantly higher channel capacity than that of conventional mesh networks.

Outage performance of cognitive radios is analyzed in this paper. The scenario under consideration requires the cognitive radio to sense whether the primary user (PU) link is free (i.e. a spectrum hole exists) before making an active transmission using that link. Multiple antennas are available at the cognitive radio link to provide array gains at the sensing stage. We derive a closed-form expression of the outage probability for cognitive transmission by classifying it into several cases. A sensing threshold is deduced according to the PU arrival model illustrated in this paper. Simulation results verify our analysis.

In this paper, we propose a new code set which has very low spectral peak to average ratio (SPAR) and good correlation properties for DS-UWB. The codes which have low SPAR are suitable for DS-UWB system which operates in UWB (3.110.4 GHz) because they can utilize more power than high SPAR codes can do. And, in order to reduce inter symbol interference (ISI) and inter piconet interferences, the codes which have good auto- and cross-correlation properties must be used. In this paper, we propose three items; (1) a new code generation method which can generate good SPAR and auto-correlation codes, (2) code selection criteria, and (3) a code set, which has been selected according to the proposed selection criteria. The proposed code set has SPAR reduced about 0.22 dB and GMF improved by 30% compared to the previous code set while it is maintaining almost same cross-correlation properties. Each code of the proposed code set, therefore, has gained 1.43 dB SIR on an average compared to that of the previous code set.

We experimentally demonstrated polarization-mode dispersion (PMD) compensation by distributing polarizers with only 1 degree of freedom (DOF) along the transmission line. The average power penalty was measured to be 0.4 dB by inserting four compensators, where average differential group delay was 47% of bit period.

We describe a way to write state machines inductively. The proposed method makes it possible to use the standard techniques for proving theorems on inductive types to verify that state machines satisfy invariant properties. A mutual exclusion protocol using a queue is used to exemplify the proposed method.

Recently, many techniques have been proposed to improve speaker identification in noise environments. Among these techniques, we consider the feature recombination technique for the multi-band approach in noise robust speaker identification. The conventional feature recombination technique is very effective in the band-limited noise condition, but in broad-band noise condition, the conventional feature recombination technique does not provide notable performance improvement compared with the full-band system. Even though the speech is corrupted by the broad-band noise, the degree of the noise corruption on each sub-band is different from each other. In the conventional feature recombination for speaker identification, all sub-band features are used to compute multi-band likelihood score, but this likelihood computation does not use a merit of multi-band approach effectively, even though the sub-band features are extracted independently. Here we propose a new technique of sub-band likelihood computation with sub-band weighting in the feature recombination method. The signal to noise ratio (SNR) is used to compute the sub-band weights. The proposed sub-band-weighted likelihood computation makes a speaker identification system more robust to noise. Experimental results show that the average error reduction rate (ERR) in various noise environments is more than 24% compared with the conventional feature recombination-based speaker identification system.

A multiple-output function can be represented by a binary decision diagram for characteristic function (BDD_for_CF). This paper presents a method to represent multiple-output incompletely specified functions using BDD_for_CFs. An algorithm to reduce the widths of BDD_for_CFs is presented. This method is useful for decomposition of incompletely specified multiple-output functions. Experimental results for radix converters, adders, a multiplier, and lists of English words show that this method is useful for the synthesis of LUT cascades. An implementation of English words list by LUT cascades and an auxiliary memory is also shown.

VoIP is one of the key technologies for recent telecommunication services. In addition to the migration from the conventional PSTN to IP networks, mobile networks will follow the PSTN in moving to an IP-based infrastructure. Due to limited radio resources, the speech bitrate in mobile networks must be more strongly compressed than is true in PSTN. This will lead to a heterogeneous network environment, in which different speech codecs are employed in fixed and mobile networks. Therefore, from the viewpoint of designing and managing the QoE (Quality of Experience) of end-to-end telephony services, establishing a method to evaluate the quality of VoIP in such a heterogeneous network environment is very important. The quality of speech communication services should be discussed in subjective terms. Subjective quality assessment is time-consuming and expensive, however, so objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. To establish an objective method to evaluate the end-to-end quality of speech in a heterogeneous network environment, this paper proposes a method for estimating the end-to-end listening quality based on the quality in each individual segment. This method is very important because conventional technologies such as the E-model, which was standardized as ITU-T Recommendation G.107, cannot accurately estimate overall quality based on segmental qualities. The experimentals show that the proposed method offers better performance in terms of quality estimation than the conventional method.

Eye tracking is a useful tool for accurately mapping where and for how long an individual learner looks at a video/image, in order to obtain immediate information regarding the distribution of a learner's attention among the elements of a video/image. This paper describes a quantitative investigation into the effect of voice navigation in web-based learning materials.

The open-vocabulary name recognition technique is one of the most challenging tasks in the application of automatic Chinese speech recognition technology. It can be used as the free name input method for telephony speech applications and automatic directory assistance systems. A Chinese name usually has two to three characters, each of which is pronounced as a single tonal syllable. Obviously, it is very confusing to recognize a three-syllable word from millions to billions of possible candidates. A novel interactive automatic-speech-recognition system is proposed to resolve this highly challenging task. This system was built as an open-vocabulary Chinese name recognition system using character-based approaches. Two important character-input speech-recognition modules were designed as backoff approaches in this system to complete the name input or to correct any misrecognized characters. Finite-state networks were compiled from regular grammar of syllable spellings and character descriptions for these two speech recognition modules. The possible candidate names cover more than five billions. This system has been tested publicly and proved a robust way to interact with the speaker. An 86.7% name recognition success rate was achieved by the interactive open-vocabulary Chinese name input system.