This paper presents a method of temporal decomposition (TD) for line spectral frequency (LSF) parameters, called "Modified Restricted Temporal Decomposition" (MRTD), and its application to low rate speech coding. The LSF parameters have not been used for TD due to the stability problems in the linear predictive coding (LPC) model. To overcome this deficiency, a refinement process is applied to the event vectors in the proposed TD method to preserve their LSF ordering property. Meanwhile, the restricted second order TD model, where only two adjacent event functions can overlap and all event functions at any time sum up to one, is utilized to reduce the computational cost of TD. In addition, based on the geometric interpretation of TD the MRTD method enforces a new property on the event functions, named the "well-shapedness" property, to model the temporal structure of speech more effectively. This paper also proposes a method for speech coding at rates around 1.2 kbps based on STRAIGHT, a high quality speech analysis-synthesis method, using MRTD. In this speech coding method, MRTD based vector quantization is used for encoding spectral information of speech. Subjective test results indicate that the speech quality of the proposed speech coding method is close to that of the 4.8 kbps FS-1016 CELP coder.

It is believed that distant-talking speech recognition in a noisy environment requires a large-scale microphone array. However, this cannot fit into small consumer devices. Our objective is to improve the performance with a limited number of microphones (preferably only left and right). In this paper, we focused on a profile that is the shape of the power distribution according to the beamforming direction. An observed profile can be decomposed into known profiles for directional sound sources and a non-directional background sound source. Evaluations confirmed this method reduced the CER (Character Error Ratio) for the dictation task by more than 20% compared to a conventional 2-channel Adaptive Spectral Subtraction beamformer in a non-reverberant environment.

In this paper, a grey filtering approach based on GM(1,1) model is proposed. Then the grey filtering is applied to speech enhancement. The fundamental idea in the proposed grey filtering is to relate estimation error of GM(1,1) model to additive noise. The simulation results indicate that the additive noise can be estimated accurately by the proposed grey filtering approach with an appropriate scaling factor. Note that the spectral subtraction approach to speech enhancement is heavily dependent on the accuracy of statistics of additive noise and that the grey filtering is able to estimate additive noise appropriately. A magnitude spectral subtraction (MSS) approach for speech enhancement is proposed where the mechanism to determine the non-speech and speech portions is not required. Two examples are provided to justify the proposed MSS approach based on grey filtering. The simulation results show that the objective of speech enhancement has been achieved by the proposed MSS approach. Besides, the proposed MSS approach is compared with HFR-based approach in [4] and ZP approach in [5]. Simulation results indicate that in most of cases HFR-based and ZP approaches outperform the proposed MSS approach in SNRimp. However, the proposed MSS approach has better subjective listening quality than HFR-based and ZP approaches.

This paper proposes a silence compression algorithm operating at multi-rates (MR) and with dual-bandwidths (DB), a narrowband and a wideband, for the MPEG (Moving Picture Experts Group)-4 CELP (Code Excited Linear Prediction) standard. The MR/DB operations are implemented by a Variable-Frame-size/Dual-Bandwidth Voice Activity Detection (VF/DB-VAD) module with bandwidth conversions of the input signal, and a Variable-Frame-size Comfort Noise Generator (VF-CNG) module. The CNG module adaptively smoothes the Root Mean Square (RMS) value of the input signal to improve the coding quality during transition periods. The algorithm also employs a Dual-Rate Discontinuous Transmission (DR-DTX) module to reduce an average transmission bitrate during silence periods. Subjective test results show that the proposed silence compression algorithm gives no degradation in coding quality for clean and noisy speech signals. These signals include about 20 to 30% non-speech frames and the average transmission bitrates are reduced by 20 to 40%. The proposed algorithm has been adopted as a part of the ISO/IEC MPEG-4 CELP version 2 standard.

One of fundamental issues on the F0 contour is modeling relationship between F0 parameters and linguistic information of a sentence. This paper proposes a stochastic F0 model which probabilistically models the relationship between the F0 contour and the linguistic information. For the application of speech synthesis, an F0 generator selects the most probable F0 contour from candidates given by a probabilistic F0 model. An F0 contour of a Japanese sentence is represented by concatenation of F0 patterns of a Japanese syntactic unit, bunsetsu. A bunsetsu F0 pattern is composed of an F0 average and an F0 shape. The F0 average is independently predicted for each bunsetsu by a quantification theory from linguistic features of the bunsetsu. The most probable sequence of bunsetsu F0 shapes for a sentence is found in the F0 shape database based on a probabilistic measure. The probability that an F0 contour is observed for a sentence is defined by two kinds of probabilities, the F0 shape production and the F0 shape bigram. The latter is a probability of adjacent occurrence of two F0 shapes, which is similar to a word bigram in speech recognition. Several typical bunsetsu F0 shapes are extracted by clustering of training data and stored in the F0 shape database. The probability of the F0 shape production is computed for each bunsetsu based on distribution of values for the linguistic feature in a cluster. The RMS prediction errors of the F0 contour are 0.26[octave].

Based on a modified step-by-step decoding procedure, a high-speed pipelined Reed-Solomon decoder is presented. The decoder requires only the delay time of three 2-input XOR gates for decoding each coded symbol. The decoder can be operated in a bit rate of Gbits/sec order and thus suitable for the very high speed data transmission systems.

This paper describes a simple, inexpensive technique for intentionally broadening and flattening the spectrum of a DC-DC converter (switching regulator) to reduce Electro-Magnetic Interference (EMI). This noise spectrum broadening technique involves intentionally introducing pseudo-random dithering of control clock timing, which can be achieved by adding simple digital circuitry. This technique can significantly reduce noise power spectrum peaks at the DC-DC converter output. For our test case circuit, measurements showed that noise power was reduced by 5.7 dBm at the main peak, by 15.6 dBm at the second peak and by 12.8 dBm at the third peak. This simple, inexpensive technique can be applied to most conventional switching regulators by adding simple digital circuitry, and without any modification of the design of other parts.

The use of fluid Generalized Processor Sharing (GPS) algorithm for integrated service networks has received much attention since early 1990's because of its desirable properties in terms of delay bound and service fairness. Many Packet Fair Queuing (PFQ) algorithms have been developed to approximate GPS. However, owing to the implementation complexity, it is difficult to support a large number of sessions with diverse service rates while maintaining the GPS properties. The grouping architecture has been proposed to dramatically reduce the implementation complexity. However, the grouping architecture can only support a fixed number of service rates, thus causing the problems of granularity, bandwidth fairness, utilization, and immunity of flows. In this paper, we propose a new implementation approach called dual-rate grouping, which can significantly alleviate the above problems. Compared with the grouping architecture, the proposed approach possesses better performance in terms of approximating per session-based PFQ algorithms without increasing the implementation complexity.

Many types of adaptive algorithms based on the MMSE criterion for co-channel interference suppression in DS/CDMA systems have been studied in great detail. However, these algorithms have such a problem that the training speed is greatly dropped under the strong near-far problem. In this paper, we propose and analyze an adaptive filter based on the Maximum Signal to Interference and Noise Ratio (MSINR) criterion, called adaptive MSINR filter. This filter is basically equivalent to the adaptive filter based on the MMSE criterion. However, due to the structual difference, the convergence speed is greatly improved. Specifically, the de-spreading vector in this filter is so renewed as to maximize the Signal to Interference and Noise Ratio (SINR) by minimizing the de-spread interference and noise power under the condition that the de-spread desired signal power keeps constant. So the proposed filter uses the estimated interference and noise signal calculated by subtracting the estimated desired signal from the received signal. It is just the reason why the adaptive MSINR filter shows remarkable convergence speed. And to satisfy the constant signal power condition, the projection matrix onto the orthogonal complement of the desired signal space is used for the de-spreading vector. For the proposed filter, we analyze the convergence modes and also investigate the de-spread interfernce and noise power for calculating the theoretical SINR curve. Then, we conduct some computer simulations in order to show the difference between this filter and the conventional one in terms of the SINR convergence speed. As the result, we confirm that the adaptive filter based on the MSINR criterion achieves significant progress in terms of the SINR convergence speed.

We propose and demonstrate a novel method to measure the polarization mode dispersion (PMD) of optical devices. The device under test (DUT) is installed in a fiber laser cavity which can operate at multiwavelength. PMD can be evaluated by the wavelength spacing of the multiwavelength laser output spectrum. In our method, the maximum extrema wavelength is easier to be identified than in the conventional fixed-analyzer (FA) method. We measure the PMD of polarization maintaining fibers (PMFs) and the ITU-T round robin KDD samples.

In wireless communication systems, low-power metrics is becoming a burdensome problem in the portable terminal design, because of portability constraints. This paper presents design architecture of a low-power Digital Matched Filter (DMF) for the direct-sequence spread-spectrum communication system such as WCDMA or wireless LAN. The proposed approach for power savings focuses on the architecture of the reception registers and the correlation-calculating unit, which dissipate the majority of the power in a DMF. The main features are asynchronous latch clock generation for the reception registers, parallelism of correlation calculation operations and bit manipulation for chip-correlation operations. A DMF is designed in compliance with the WCDMA specifications incorporating the proposed techniques, and its properties are evaluated by computer simulations at the gate level using 0.18-µm CMOS standard cell array technology. As a result, the power consumption of the proposed DMF is estimated to be 9.3 mW (@15.6 MHz, 1.6 V), which is below 40% of the power consumed by a general DMF.

This paper proposes a multipath interference canceller (MPIC) employing multipath interference (MPI) replica generation (MIG) utilizing previously transmitted packet combining (PTPC), which is well-suited to incremental redundancy, in order to achieve a peak throughput of nearly 8 Mbps in a multipath fading environment in high-speed downlink packet access (HSDPA). In our scheme, more accurate MPI replica generation is possible by generating MPI replicas utilizing the soft-decision symbol sequence of the previously transmitted packets in addition to that of the latest transmitted packet. Computer simulation results elucidate that the achievable throughput of the MPIC employing MIG-PTPC is increased by approximately 100 kbps and 200 kbps and the required average received signal energy per symbol-to-background noise power spectrum density ratio (Es/N0) per antenna at the throughput of 0.8 normalized by the maximum throughput is improved by about 0.3 and 0.7 dB compared to that of the MPIC using the soft-decision symbol sequence after Rake combining of the last transmitted packet both in 2- and 3-path Rayleigh fading channels for QPSK and 16QAM data modulations, respectively. Furthermore, we clarify that the maximum peak throughput using the proposed MPIC with MIG-PTPC coupled with incremental redundancy achieves approximately 7 Mbps and 8 Mbps with 16QAM and 64QAM data modulations in a 2-path Rayleigh fading channel, respectively, within a 5-MHz bandwidth.

In an order-specified multisignature scheme, one can verify not only a set of signers who have signed the message but also its signing order. Though we have seen several schemes with such properties proposed, none of them is given the security proof against active adversaries. The scheme can be easily modified to be an order-specified multisignature scheme, but still has the restriction that the possible signing orders are only ones of the type of serial signing. In this paper, we propose the first order-specified multisignature scheme, which is shown to be secure against adaptive chosen-message insider attacks in the random oracle model, and which allows the signing orders to form like any series-parallel graphs. The security is shown by using ID-reduction technique, which reduces the security of multisignature schemes to those of multi-round identification schemes. Furthermore, we discuss the efficiency of the proposed scheme and the upper bound of the possible number of participating signers.

This paper discusses two types of polyphase sequence sets, which will successfully provide CDMA systems without co-channel interference. One is a type of ZCZ sets, whose periodic auto-correlation functions take zero at continuous shifts on both side of the zero-shift, and periodic cross-ones also take zero at the continuous shifts and the zero-shift. The other is a new type of sets consisting of some subsets of polyphase sequences with zero cross-correlation zone, called ZCCZ sets, whose periodic cross-correlation functions among different subsets have take zero at continuous shifts on both side of the zero-shift including the zero-shift. The former can achieve a mathematical bound, and the latter can have large size.

There is a strong demand to expand captioned broadcasting for TV news programs in Japan. However, keyboard entry of captioned manuscripts for news program cannot keep pace with the speed of speech, because in the case of Japanese it takes time to select the correct characters from among homonyms. In order to implement simultaneous subtitled broadcasting for Japanese news programs, a simultaneous subtitling system by speech recognition has been developed. This system consists of a real-time speech recognition system to handle broadcast news transcription and a recognition-error correction system that manually corrects mistakes in the recognition result with short delay time. NHK started simultaneous subtitled broadcasting for the news program "News 7" on the evening of March 27, 2000.

We investigate dispersion compensation using dispersion-compensating fibers (DCFs) for ultrashort light pulse code division multiple access (CDMA) communication systems in a multi-user environment. We employ fiber link that consists of a standard single-mode fiber (SMF) connected with two different types of DCFs. Fiber dispersion can be effectively decreased by adjusting the length ratios of DCFs to SMF appropriately. Some criteria for dispersion compensation are proposed and their performances are compared. We theoretically derive a bit error rate (BER) of ultrashort light pulse CDMA systems including the effects of the dispersion and multiple access interference (MAI). Moreover, we reveal the mutual relations among BER performance, fiber dispersion, MAI, the number of chips, a bandwidth of a signal, and a transmission distance for the first time. As a result, we show that our compensation strategy improves system performance drastically.

A wireless communications system with a transmission speed of 18 Mbit/s is presented using the 2.4 GHz ISM band. This system employs the Carrier Frequency Offset-Spread Spectrum (CFO-SS) scheme and the Dual-Polarization Staggered Transmission (DPST) scheme. The 18 Mbit/s CFO-SS system (named CFO-SS18) was developed and its performance evaluated in fields. In this paper, the detailed operating principle of CFO-SS and DPST schemes, together with the specifications and structures of CFO-SS18, are presented. Results of indoor and field tests obtained by using CFO-SS18 are also presented.

In this paper, we propose and describe a new synchronizer for the FFT timing applicable to multi-carrier spread-spectrum (MC-SS) communication systems. The performance of the synchronizer is evaluated in terms of false- and miss-detection probabilities in the presence of additive white Gaussian noise (AWGN) and Rayleigh fading.

Frequency modulation (FM) noise spectroscopy with diode laser is applied to high-resolution Doppler-free spectroscopy of Cs atomic vapor near a dielectric surface with evanescent-wave pump-probe configuration. Both high resolution and high sensitivity are realized by using an extremely simple experimental setup, in which no sweep or precise tuning of laser frequency are required. Several experimental configurations of optical near-field spectroscopy are demonstrated, which is useful for an extensive study of resonant interactions of atoms and microscopic electronic systems in optical near-fields.

We calculated linear and nonlinear responses of a Kerr nonlinear microsphere sandwiched by two prisms using the excitation of whispering gallery modes due to near-field coupling. As numerical calculations, the finite-difference time-domain method that takes into account the Kerr nonlinear effect was used. We dealt with two types of spheres, i.e., the Kerr-material sphere and the dielectric sphere coated by the Kerr material. It was found that the optical switching phenomena are induced in such spheres. The switching results from the fact that the variations of the refractive index of the nonlinear spheres affect the excitation condition of the whispering gallery modes.