献给您一个多彩的世界：25台显示器比较评测报告

[编者按]电子工业部计算机与微电子发展中心上海测试发布展示中心经电子工业部办公厅批准成立，由电子工业部计算机与微电子发展研究中心（CCID）主办，是国内第一个计算机软硬件测试发布展示中。C。该中。C以CCID及其所属的中国计算机报社测试实验室（CIWLab）和中国软件评测中心（CSTC）先进的评测技术与工艺为依托，借鉴IDG的技术与方法以及国内外资深测试专家的经验，以科学的脚踏实地的工作作风和“忠实、公正。廉洁、权威”的工作原则，对国内市场上的PC类计算机软硬件、外设、网络产品、银行机具及其他微电子产品在类似用…     

入侵检测系统评测研究进展(上)

虽然入侵检测技术已经成为信息安全体系的重要组成部分,然而到目前为止,还没有被广泛认同的入侵检测系统评测标准。用户和研究人员时常对入侵检测系统和新的检测算法的有效性抱有疑问,解决这些问题的关键在于形成完善的入侵检测系统评测方法学。通过评测结果,可以分析现有技术的不足,从而为IDS技术的进一步研究和发展提供指导。本文对已有的IDS测试和评测研究工作分类进行了介绍,并作了相应的比较和分析。     

PC LABS 最新的基准测试内幕

在1997年底的系统的图形适配器的评测（’97和’98年度最佳PC）中，PC Magozine Labs引入了它已成为工业标准的PC基准测试程度的最新版本。Winstone 98,Version1.0；WinBench 98，Version 1.0以及3D WinBench98,Version 1.0。这些测试工具反映了     

高性能计算平台基于HPCC的性能分析

HPCC（High Performance Computing Challenge）基准较Linpack能够更全面反映高性能系统性能。但是HPCC测试结果是若干个指标项,缺少一个整体的,直观而统一的评价结果,一直未能被广泛地接受。使用HPCC测试集对两个高性能平台进行了性能评测,并在此基础上提出了一种简单易行的HPCC测试数据分析处理方案对HPCC测试结果进行分析,得到一个直观而统一的HPCC的测试结果。该结果清晰地反映出每个系统的优势和不足,并且依据该分析结果对两个高性能计算平台的性能进行了比较。     

掌纹鉴别自动性能评估

研究关于掌纹鉴别系统的性能评测问题,设计并实现了一个用于性能评测的自动测试系统（ASPE）,此系统采用的是掌纹识别系统的处理核心。同时还提出了一种基于特征点（终点和分叉点）的评测方法和用于量化评估的俩个主要指标-召回率和准确率.此系统可以直接在样张上显示标准特征点和提取特征点,方便开发人员的对比并发现问题,从速度和精确度的测试结果看出自动的性能评测在效率和准确上都大大超过了人工测试。     

基于DFA的多级IDS安全性评测

本文在对IDS进行安全性评测的基础上,提出了基于DFA的多级混合评测的方法,使评测过程更简单直观,实验证明这种方法提高了原有测试系统的广度与深度,使测试结果更加可信。     

基于实体名的文本自动综述研究

自动文摘是自然语言处理的一个重要分支，在信息检索领域中有着重要的用途．文本自动综述是自动文摘在多文档上的推广。本文提出了基于实体名扩展的自动综述方法，这种方法认为综述中的实体名个数反映其中所蕴含信，S量的多少。我们用该方法实现针对事件的自动综述生成，并参加了2003年文本理解会议(Document Understanding Conference，DUC)进行统一评测，DUC反馈的评测结果显示这种方法是有效的。此外，本文还对文本理解会议的任务、评测方法和测试结果做了简单介绍。     

权威测试 指定平台

在IT尤其是PC核心技术、规范、产品更新速度不断加快情况下，如何保证产品评测的客观性、公正性、可比性，是当前所有从事评测的组织、机构，甚至每个购买者都必须面对的问题。从1994年《个人电脑》实验室在中国刚刚建立的那时起，我们就一直力求在所有的产品评测当中贯彻这二个原则。为了更好地维系一个统一的平台、统一的评价体系、统一的测试系统，我们与各个产品领域中拥有领先技术优势和产品研发能力的厂商建立合作关系，共同搭建《个人电脑》实验室的各项测试平台。     

汉语自动分词和词性标注评测

本文介绍了2003年“863中文与接口技术”汉语自动分词与词性标注一体化评测的一些基本情况,主要包括评测的内容、评测方法、测试试题的选择与产生、测试指标以及测试结果,并对参评系统的切分和标注错误进行了总结。文中着重介绍了测试中所采用的一种柔性化的自动测试方法,该方法在一定程度上克服了界定一个具体分词单位的困难。同时,对评测的结果进行了一些分析,对今后的评测提出了一些建议。     

Web服务器性能评测

Web服务器性能评测是一种理解Web服务器对不同负载反应能力的方法，它对Web服务器的容量规划和性能增强有很大的帮助。讨论了Web服务器性能评测的原理、方法、难点及解决方案，介绍了基于Web负载的特点、ON／OFF源模型及浏览器／服务器体系结构，开发了一个Web服务器性能评测工具－WSBench。WSBench产生渐近自相似的HTTP请求序列，从静态文档、动态文档（没有数据库存取）、动态文档（有数据库存取）及前三者根据Zipf法则的组合4个层次来评测Web服务器的性能。性能测试结果表现为每秒请求数、每秒字节数和往返时间3个指标。最后讨论了Web服务器性能问题及使用WSBench测得的指标来建议Web服务器性能增强可以采用的方法。     

The emotional quality of speech in voice services

The emotional quality of speech is defined as the global qualitative and hedonic impressions experienced by listeners. This research investigated the emotional quality of speech samples used in voice services. In a first experiment listening tests were conducted using 200 messages generated by 20 female speakers who pronounced two sentences in five elocution styles. Listeners grouped the messages according to similarities in terms of the impression of the messages. Verbal comments regarding hedonic effect on the listener and acoustic parameters of the voices' timbre and intonation were analysed. In a second experiment, the 200 messages were evaluated according to 20 criteria extracted from the first experiment. The results produced a precise perceptive portrait for each sequence, giving a full picture of the listeners' impressions of what they heard. The results can be applied to the design of voice services, as was done for the voicemail of France Telecom Orange.     

Robust processing techniques for voice conversion

Differences in speaker characteristics, recording conditions, and signal processing algorithms affect output quality in voice conversion systems. This study focuses on formulating robust techniques for a codebook mapping based voice conversion algorithm. Three different methods are used to improve voice conversion performance: confidence measures, pre-emphasis, and spectral equalization. Analysis is performed for each method and the implementation details are discussed. The first method employs confidence measures in the training stage to eliminate problematic pairs of source and target speech units that might result from possible misalignments, speaking style differences or pronunciation variations. Four confidence measures are developed based on the spectral distance, fundamental frequency (f₀) distance, energy distance, and duration distance between the source and target speech units. The second method focuses on the importance of pre-emphasis in line-spectral frequency (LSF) based vocal tract modeling and transformation. The last method, spectral equalization, is aimed at reducing the differences in the source and target long-term spectra when the source and target recording conditions are significantly different. The voice conversion algorithm that employs the proposed techniques is compared with the baseline voice conversion algorithm with objective tests as well as three subjective listening tests. First, similarity to the target voice is evaluated in a subjective listening test and it is shown that the proposed algorithm improves similarity to the target voice by 23.0%. An ABX test is performed and the proposed algorithm is preferred over the baseline algorithm by 76.4%. In the third test, the two algorithms are compared in terms of the subjective quality of the voice conversion output. The proposed algorithm improves the subjective output quality by 46.8% in terms of mean opinion score (MOS).     

The emotional quality of speech in voice services

The emotional quality of speech is defined as the global qualitative and hedonic impressions experienced by listeners. This research investigated the emotional quality of speech samples used in voice services. In a first experiment listening tests were conducted using 200 messages generated by 20 female speakers who pronounced two sentences in five elocution styles. Listeners grouped the messages according to similarities in terms of the impression of the messages. Verbal comments regarding hedonic effect on the listener and acoustic parameters of the voices' timbre and intonation were analysed. In a second experiment, the 200 messages were evaluated according to 20 criteria extracted from the first experiment. The results produced a precise perceptive portrait for each sequence, giving a full picture of the listeners' impressions of what they heard. The results can be applied to the design of voice services, as was done for the voicemail of France Telecom Orange.     

A HMM-based Mandarin Chinese Singing Voice Synthesis System

We propose a mandarin Chinese singing voice synthesis system, in which hidden Markov model (HMM)-based speech synthesis technique is used. A mandarin Chinese singing voice corpus is recorded and musical contextual features are well designed for training. F0 and spectrum of singing voice are simultaneously modeled with context-dependent HMMs. There is a new problem, F0 of singing voice is always sparse because of large amount of context, i.e., tempo and pitch of note, key, time signature and etc. So the features hardly ever appeared in the training data cannot be well obtained. To address this problem, difference between F0 of singing voice and that of musical score (DF0) is modeled by a single Viterbi training. To overcome the over-smoothing of the generated F0 contour, syllable level F0 model based on discrete cosine transforms (DCT) is applied, F0 contour is generated by integrating two-level statistical models. The experimental results demonstrate that the proposed system outperforms the baseline system in both objective and subjective evaluations. The proposed system can generate a more natural F0 contour. Furthermore, the syllable level F0 model can make singing voice more expressive.      

Transforming Perceived Vocal Effort and Breathiness Using Adaptive Pre-Emphasis Linear Prediction

This paper presents a technique to transform high-effort voices into breathy voices using adaptive pre-emphasis linear prediction (APLP). The primary benefit of this technique is that it estimates a spectral emphasis filter that can be used to manipulate the perceived vocal effort. The other benefit of APLP is that it estimates a formant filter that is more consistent across varying voice qualities. This paper describes how constant pre-emphasis linear prediction (LP) estimates a voice source with a constant spectral envelope even though the spectral envelope of the true voice source varies over time. A listening experiment demonstrates how differences in vocal effort and breathiness are audible in the formant filter estimated by constant pre-emphasis LP. APLP is presented as a technique to estimate a spectral emphasis filter that captures the combined influence of the glottal source and the vocal tract upon the spectral envelope of the voice. A final listening experiment demonstrates how APLP can be used to effectively transform high-effort voices into breathy voices. The techniques presented here are relevant to researchers in voice conversion, voice quality, singing, and emotion.     

Measuring speech quality for text-to-speech systems: development and assessment of a modified mean opinion score (MOS) scale

The quality of text-to-speech systems can be effectively assessed only on the basis of reliable and valid listening tests to assess overall system performance. A mean opinion scale (MOS) has been the recommended measure of synthesized speech quality [ITU-T Recommendation P.85, 1994. Telephone transmission quality subjective opinion tests. A method for subjective performance assessment of the quality of speech voice output devices]. We assessed this MOS scale and developed and tested a modified measure of speech quality. This modified measure has new items specific to text-to-speech systems. Our research was motivated by the lack of clear evidence of the conceptual content of as well as the psychometric properties of the MOS scale. We present conceptual arguments and empirical evidence for the reliability and validity of a modified scale. Moreover, we employ state of the art psychometric techniques such as confirmatory factor analysis to provide strong tests of psychometric properties. This modified scale is better suited to appraise synthesis systems since it includes items that are specific to the artifacts found in synthesized speech. We believe that the speech synthesis research communities will find this modified scale a better fit for listening tests to assess synthesized speech.     

基于HLA的语音通信模拟系统

依据高层体系结构标准,通过分析语音通信模拟系统的体系结构,定义联邦成员的仿真对象模型,调用运行时间框架的API实现联邦成员的交互,根据语音通信实时仿真的特点,从应用层的角度提高语音通信模拟系统实时性,运用声明管理和数据分发管理进行数据过滤,设置全局时钟进行联邦成员间地时间同步。实验结果表明,采用的方法能够满足语音通信模拟系统的开发需求。     

水声语音通信体验质量的实时测量方法

水声语音通信质量的实时测量是保障通信质量的重要环节,利用实时测量结果可及时调整语音业务的调制参数,提高链路的自适应能力。本文提出了一种基于参数表示的语音通信质量实时估测模型,该模型提取语音3个特征参数,即：Mel频率倒谱系数（Mel-frequency cepstum coefficient,MFCC）、线性预测倒谱系数（Linear predictive cepstrum coefficient,LPCC）及加权对数谱（Log spectral deviation,LSD）,构建3种特征参数的权重谱失真测度。利用失真测度与接收语音质（Perceptual evaluation of speech quality-mean opinion score,PESQ-MOS）之间的映射关系,建立语音质量估测模型。引入动态MFCC（Dynamic Mel-frequency cepstrtum coefficient,DMFCC）的谱失真测度作为质量估测模型的调节因子,使估测系统具有更好的适应性。实验及海测结果显示,利用本模型估测的语音MOS值与人主观感受误差较小,具有一定的实用性。     

基于GMM的歌曲Morphing算法

针对业余歌手模仿专业歌手唱歌过程中音色不变的问题,提出一种基于高斯混合模型(GMM)的中文歌曲Morphing算法,采用GMM对语音频谱建模,并通过混合业余歌手和专业歌手的语音频谱,实现歌曲的音色转换。结果显示,混合比例因子k=0或1时,ABX测试正确率均为100%,0相似文献   

基于E-Model的语音帧分组传输性能研究

voIP的语音帧分组大小是实时语音传输的关键参数。为提高网络效率和最大话路数,采用EModel的方法分析了RTP包中语音帧个数、语音长度、丢包概率和抖动缓冲区大小对语音质量的影响,给出了不同带宽时的最佳传输分组大小。仿真结果表明,在保证最基本的话音质量情况下,为不同链路确定合适的分组语音帧数能有效提高链路的最大话路数。