用Verilog HDL设计数字电路过程中的两个问题

Verilog HDL广泛应用于数字电路设计，但在实际电路设计过程中也存在一些问题.主要对语句执行顺序和建模问题进行分析讨论，在建模问题中较详细地讨论了十六位乘法器建模、行为描述建模、状态建模和30 s建模.在此基础上，指出了解决这两个重要问题的关键是明白设计要求，看清电路自身特点，抓住问题的关键点.此研究为今后用Verilog HDL设计数据电路的工作提供了一定借鉴.     

Banach空间中几乎渐进拟非扩张型映像的具混合误差的迭代逼近

在Banach空间中引入了一类新的几乎渐进拟非扩张映像，人们熟知的非扩张映像类、渐进非扩张映像类以及渐进非扩张型映像类都是这种映像的特例．本文研究了用于逼近几乎渐进拟非扩张型映像的具混合误差的修改了的Ishikawa迭代序列收敛性问题，并给出了此迭代序列收敛到不动点的充分必要条件．本文的结果推广了ChangSS等人的最新结果．     

分组轮回扫描法的摩托车电喇叭耐久检测

为了实现大批量摩托车电喇叭耐久性检测，根据摩托车电喇叭耐久检测的国家标准及用户要求，提出了分组轮回扫描检测电喇叭耐久性的方法，用单驱动电源依次轮回驱动，采用二个电源分组实现一批次8只电喇叭的耐久性检测，研制出了适用于两种标称电压、不同规格尺寸的摩托车电喇叭的耐久性试验台，阐述了系统的工作原理和故障判别模式．所设计试验台减少了驱动电源数目，简化了控制系统，降低了系统成本。     

主变保护误动作一例分析及探讨

2004年7月20日,110kV长城变#1主变因差动保护动作而造成主变三侧开关跳闸,经分析是由于主变保护差动回路被误短接所致,为此需采取相关的技术措施和组织措施,防止类似事故的发生.     

循环水系统海生物污染控制

分析珠海电厂循环水冷却系统海生物污染情况，探讨控制循环水海生物污染的方法，介绍用于控制海生物生长的杀生剂CT1300的特点和应用经验，并对CT1300与氯气杀灭循环水冷却系统海生物的实效进行了比较。     

三相PWM逆变器软开关动作时序分析

通过对软开关三相PWM逆变器的研究和实验，针对在实验中出现谐振槽不正常的现象，分析了本软开关电路拓扑下实现软开关动作的工作原理及谐振动作时序，阐述了PWM驱动信号与谐振电路的动作时序间的关系，指出要使谐振能正常进行，必须要使PWM驱动信号的开通动作控制在谐振电感的电流iLr变为负之前，这样才能在直流母线电压上出现一个完整的谐振槽，保证逆变器主电路开关管实现软开关动作。     

管理信息系统中流水号生成方法与实现

在管理信息系统中，生成、控制和管理流水号十分重要。基于Oracle数据库系统，讨论了序号生成器sequence的基本用法，描述了一种产生连续或非连续流水号的优化方法及实现步骤，同时扩展了序号生成器的用法。结合具体流水号数据的特点和存放格式，给出了PowerBuilder的生成序列号函数实例、主要程序代码以及调用方法。在实际应用中，某大型管理信息系统采用了这种设计方法，取得了较好的效果。     

Practical crack control during the construction of precast segmental box girder bridges

Cracks that occurred in the bottom slab of a precast segmental bridge were investigated through a construction sequence analysis, which revealed that the cracks were caused by excessive deformation during temporary post-tensioning while joining the segments. In addition, a parametric study was performed to evaluate the effects of the prestressing sequence, bottom slab thickness, and position of the prestressing anchors. The structural behavior of the girder sections was greatly affected by the thickness of the bottom slab and the position of prestressing anchors, but not by the prestressing sequence. Based on the results, a construction method that prevents the cracks is proposed.     

An algorithm for similar utterance section extraction for managing spoken documents

This paper proposes a new, efficient algorithm for extracting similar sections between two time sequence data sets. The algorithm, called Relay Continuous Dynamic Programming (Relay CDP), realizes fast matching between arbitrary sections in the reference pattern and the input pattern and enables the extraction of similar sections in a frame synchronous manner. In addition, Relay CDP is extended to two types of applications that handle spoken documents. The first application is the extraction of repeated utterances in a presentation or a news speech because repeated utterances are assumed to be important parts of the speech. These repeated utterances can be regarded as labels for information retrieval. The second application is flexible spoken document retrieval. A phonetic model is introduced to cope with the speech of different speakers. The new algorithm allows a user to query by natural utterance and searches spoken documents for any partial matches to the query utterance. We present herein a detailed explanation of Relay CDP and the experimental results for the extraction of similar sections and report results for two applications using Relay CDP. Yoshiaki Itoh has been an associate professor in the Faculty of Software and Information Science at Iwate Prefectural University, Iwate, Japan, since 2001. He received the B.E. degree, M.E. degree, and Dr. Eng. from Tokyo University, Tokyo, in 1987, 1989, and 1999, respectively. From 1989 to 2001 he was a researcher and a staff member of Kawasaki Steel Corporation, Tokyo and Okayama. From 1992 to 1994 he transferred as a researcher to Real World Computing Partnership, Tsukuba, Japan. Dr. Itoh's research interests include spoken document processing without recognition, audio and video retrieval, and real-time human communication systems. He is a member of ISCA, Acoustical Society of Japan, Institute of Electronics, Information and Communication Engineers, Information Processing Society of Japan, and Japan Society of Artificial Intelligence. Kazuyo Tanaka has been a professor at the University of Tsukuba, Tsukuba, Japan, since 2002. He received the B.E. degree from Yokohama National University, Yokohama, Japan, in 1970, and the Dr. Eng. degree from Tohoku University, Sendai, Japan, in 1984. From 1971 to 2002 he was research officer of Electrotechnical Laboratory (ETL), Tsukuba, Japan, and the National Institute of Advanced Science and Technology (AIST), Tsukuba, Japan, where he was working on speech analysis, synthesis, recognition, and understanding, and also served as chief of the speech processing section. His current interests include digital signal processing, spoken document processing, and human information processing. He is a member of IEEE, ISCA, Acoustical Society of Japan, Institute of Electronics, Information and Communication Engineers, and Japan Society of Artificial Intelligence. Shi-Wook Lee received the B.E. degree and M.E. degree from Yeungnam University, Korea and Ph.D. degree from the University of Tokyo in 1995, 1997, and 2001, respectively. Since 2001 he has been working in the Research Group of Speech and Auditory Signal Processing, the National Institute of Advanced Science and Technology (AIST), Tsukuba, Japan, as a postdoctoral fellow. His research interests include spoken document processing, speech recognition, and understanding.     

Robust multi bit and high quality audio watermarking using pseudo-random sequences

A robust multi bit and high quality audio watermarking technique in time domain is proposed in this paper. Watermarking is a technique used to label digital media by hiding copyright or other information into the underlying data. The watermark must be imperceptible and undetectable by the user and should be robust to various types of distortion. To enhance the robustness and taper-resistance of the embedded watermark, in this paper a multi bit technique is employed. Instead of embedding one bit into an audio frame, multiple bits can be embedded into each audio sub-frame. For attackers, since they do not know the parameters, this significantly reduces the possibility of unauthorized bit detection and removal of watermark. In real-time watermarking applications, robustness is not the only factor that plays an important role. The other factor that plays a very an important role is computational complexity. In general, audio file is transmitted in compressed form. Real-time watermark embedding must take into account this compressed form, because first decompressing the data, adding a watermark and then recompressing the data is computationally too demanding. In this paper, we propose robust and low complexity audio watermarking algorithm. To evaluate the performance of the proposed audio watermarking algorithm, subjective and objective quality tests including bit error rate (BER) and signal-to-noise ratio (SNR) were conducted. Compared to traditional one bit embedding algorithm, the proposed algorithm gives highly recovery rate after attack by commonly used audio data manipulations, such as low pass filtering, requantization, resampling, and MP3 compression.