对新型宽带语音编解码器AMR-WB的研究

文章介绍了一种应用于第三代移动通讯的新型语音编解码器,该编解码器的语音带宽拓展为50Hz到7000Hz,编码后语音的自然度很高,用在3G移动通讯系统的多媒体服务、宽带包交换网络、音频和视频会议等等。在简单介绍该语音编解码器的产生背景后,着重介绍了该语音编解码器组成和特点。     

新型宽带语音编解码器AMR-WB的语音端点检测(VAD)研究

AMR-WB是一种应用于3G上的一种新型宽带语音编解码器,该编解码器将语音带宽拓展到7kHz,采样频率为16kHz,编码后的语音在自然度、音乐方面有较大的改善,将在3G移动通信系统、ISDN上的高保真电话、数字无线广播等各个方面有广泛的应用.文章着重介绍该语音编解码器中有关语音端点检测(VAD)算法,并给出了测试结果.     

G.723.1语音编解码器优化策略的研究

在语音编解码器的实际应用中,语音编解码器的编解码时间是一个关键的性能指标。本文通过对G.723.1语音编解码器客观测试和使用VC^＋＋的工具profile对G.723.1语音编解码器测试结果的分析,提出对G.723.1语音编解码器的优化应着重于对计算复杂度较大的模块进行优化。如基音估计、自适应码本搜索、固定码本搜索等模块。并针对这些模块给出相应的优化方法。最后对根据优化策略优化的代码和未优化代码进行了测试,结果显示优化代码比未优化代码的运行时间减少了24％－31％。     

基于Android的移动视频监控客户端

近年来,移动通讯网络的迅速发展和Android系统智能手机的普及使得移动视频监控成为一个重要的研究方向. 在深入研究流媒体传输技术、视频编解码技术和SIP协议的基础上,设计了一种具有语音通话功能的移动视频监控系统,并重点对基于Android平台的移动视频监控客户端进行研究与设计.     

关于第三代通信系统的探索

移动通信是世界电信领域中发展最迅速,前景最广阔的技术,是通讯技术向信息时代全面发展的一个重要标志。建立在全球移动综合业务基础上的第三代通信系统(3rd Generation,简称为3G),为移动多媒体带来高速的数据传输带宽、完善的移动终端,同时提供多种优质的语音和非语音业务,是未来电信领域的发展方向,该文将对第三代通信系统的发展概况、结构、定位等做粗浅探索。     

捷波朗UC VOICE250耳麦 畅享统一通讯

统一通讯在某些办公领域一直是一个时髦的话题,这个英文简称为UC的概念是指将一些实时通讯服务,例如即时消息（语音聊天）、电话会议、电话（包含IP电话）,以及一些非实时通讯服务,例如语音电子邮件等整合在一起。这套系统虽然看似时髦,但是最终还是需要终端语音输入输出设备的支持,捷波朗UCVOICE250就是一款服务于统一通讯系统的耳麦。     

移动互联时代的语音识别技术

语音识别是新一代人机界面中最重要的技术之一。上一世纪最后十年,由于语音识别技术的快速发展,人们对此寄托了无限的期望和遐想。然而,至少到目前为止,从听写机到电话语音识别,语音识别的应用尚欠火候。是技术上不可逾越的障碍,还是应用不合时宜、时机?这一切将随着移动无线互联网以及新一代移动通讯的发展而产生变化。 中国科学院自动化研究所研究员、博士生导师、国家模式识别实验室副主任徐波老师将从现有语音技术、产品出发,讨论该技术未来几年在嵌入式系统、无线互联网以及语音翻译中的应用以及对信息产业可能产生的深刻影响。     

一种基于SoPC技术的iLBC语音编解码器设计实现

基于VoIP技术的语音通信发展迅速,单芯片VoIP处理器的设计方法成为当前的研究热点.iLBC作为专为窄带通信而设计的VoIP语音编解码器,可以在丢包率和延迟较高的网络环境中保持良好的语音通话质量,具有广泛的应用前景.传统的基于DSP处理器实现方法具有芯片面积大、功耗高等缺点,难以满足VoIP系统集成度高、低功耗和易于升级等需求.本文提出了一种基于SoPC技术的iLBC语音编解码器实现方案,并对自相关计算算法进行了并行计算硬件IP核设计,提高了系统的集成度、计算性能和可扩展性.理论分析和实验结果表明并行自相关计算结构有效减少了访存次数,可以获得接近30的加速比.     

G.723.1语音编码器在Blackfin平台上的实时实现①

ITU-TG.723.1是一种用于多媒体通信的双码率语音编码标准,几乎在所有的语音网关设备上面g723.1音频编解码器都是必须支持的一个标准编解码器。针对G.723.1音频编解码算法尚未在BF532＋uClinux平台上实时实现的情况,基于BF532＋uClinux平台提出了该算法实时实现的优化方案。方案减少了编解码的时延,降低了算法的复杂度,编解码整体性能提升约10倍,满足了BF532＋uClinux平台的实时性要求,并全部通过ITU测试向量的测试。最后将优化好的G723.1编解码器应用到嵌入式语音网关中,实验表明语音通话效果良好。     

语音使能的多模态输入Web应用系统设计

主要研究了语音使能的多模态输入Web应用的系统架构和运行机制。该系统在传统Web交互模式的基础上引入了分布式语音识别功能模块，使手持移动设备的用户可以同时使用语音和其他传统输入方式获得无线因特网的信息服务。     

IP电话系统中语音芯片的DSP实现

介绍了一种具有G．723．1语音编解码、自适应回声抵消DTMF编解码功能的语音芯片，以满足IP电话系统中低速率语音通信的需要。根据DSP内核的特点，对G．723．1算法进行了优化，降低了算法 复杂度；回声抵消部分使用归一化最小二乘（NL MS）自适应滤波器实现,采用Goertzel算法实现DTMF的检测。使用较少的资源,在一块TMS320C5402芯片上集成了以上全部功能。     

Performance of GSM networks with general packet radio services

Recently, most of the mobile network providers start to introduce general packet radio service (GPRS) in their existing GSM networks. GPRS is the technology that will enable more efficient Internet applications to run on mobile networks even before the installation of 3G systems. However, it is not yet clearly understood, how the new data services will affect the overall network performance. This paper provides a framework for analytical performance evaluation of a single GSM/GPRS cell based on a multidimensional Markov chain model. Important performance measures like new call and handover blocking probabilities, moments of the blocking period distributions, achievable average data rates and resource utilization are determined. Introducing a new connection admission control (CAC) algorithm, different partitioning strategies between GSM and GPRS resources are investigated. Finally, the influence of typical GPRS applications like Internet browsing on traditional GSM services has been studied.     

复杂无向网络连通性的一种高效判定算法

通信网络的拓扑结构连通性是多智能体系统一致性控制或编队控制等的理论前提.以往, 各种多智能体系统一致性控制或编队控制方面的文献仅侧重于控制协议、智能体动力学模型和控制律设计, 而缺乏对多智能体通信网络拓扑结构的连通性研究.网络连通性高效判定算法不仅是大规模多智能体系统一致性控制或编队控制的保证, 而且在图论、现代移动通信、计算机与交通等各种网络中有着重要和广泛的应用.针对复杂无向网络的连通性问题, 本文给出了一种新的高效判定算法、以及该算法的时间复杂度和空间复杂度的上界.该算法具有非常低的时间复杂度和空间复杂度, 且便于计算机实现, 因而具有重要的理论意义和广泛的实用价值.     

A robust method for soft IP handover

Multimedia streaming over wireless networks - often called mobile multimedia streaming lets users access music, movie, and news services at any time, regardless of location. Given that multimedia streaming is a key goal of third-generation and future wireless networks, vendors will soon deploy streaming clients in advanced mobile terminals. Current mobile terminals, however, fail to adequately support mobile multimedia communication because wireless networks have high packet-loss rates. To eliminate packet loss during handover, we use a packet path diversity scheme and an end-to-end bicasting mechanism that enables soft IP handover. To offset wireless errors, we use a forward error correction (FEC) scheme and embed it in the bicasting mechanism. Our bicasting method encodes the data stream and then splits it, providing more effective diversity than general bicasting, which sends the same data down both paths.' To support our method, we propose the mobile multimedia streaming protocol (MMSP), a new transport-layer protocol that supports multihoming and bicasting in combination with FEC.     

移动IP技术及其在校园网中的应用

移动IP技术是在传统网络中实现下一代网络应用的核心技术,是IP技术发展的新领域,是无线通信技术和IP技术的相互融合。本文首先描述了移动IP技术的历史与现状、移动IP的工作过程,随后提出了移动IP存在的主要问题,并介绍了相应解决方案。最后,还分析了移动IP在校园网中的应用,指出移动IP技术将在移动通信中发挥重要作用并将为今后各种新业务的开展提供前所未有的新天地。     

Advanced seamless vertical handoff architecture for WiMAX and WiFi heterogeneous networks with QoS guarantees

The development of wireless networks brings people great convenience. More state-of-the-art communication protocols of wireless networks are getting mature. People attach more importance to the connections between heterogeneous wireless networks as well as the transparency of transmission quality guarantees. In this paper, WiFi wireless networks and WiMAX mobile communication system are selected as our research subjects. VHTC (Vertical Handoff Translation Center Architecture) is proposed to improve the transmission QoS guarantees. Thus, the quality of transmission cannot be reduced enormously even by the effect of vertical handoff between heterogeneous wireless networks. This paper includes new approaches and architecture among Packet Translation, QoS Mapping, Bandwidth Borrowing Management and Vertical Handoff Protocol with a view to achieve the goal, that is, an advanced seamless heterogeneous wireless networks. Also, through the modification of WiFi module of TKN and WiMAX module of NIST, these two heterogeneous wireless networks can be connected and embedded in NS2 (Network Simulator version 2) simulation system. Afterwards, the methods, proposed in this paper, can be implemented in such simulation system. Finally, lots of satisfactory simulation results about throughput, delay, jitter and packet loss rate were given to show the brilliant performance of vertical handoff implemented by VHTC.     

一个新的移动Agent的可靠通信算法

移动Agent技术是近年来分布式计算领域中一项新兴的技术。其中有很多领域的研究尚不成熟，如Agent的可靠通信。Agent的可靠通信一直是Agent技术的一个难题，在现有的很多Agent系统中都未得到解决。比较了几种移动Agent可靠通信的算法，并提出了一种新的移动Agent的可靠通信算法。     

Minimization of communication cost through caching in mobileenvironments

Users of mobile computers will soon have online access to a large number of databases via wireless networks. Because of limited bandwidth, wireless communication is more expensive than wire communication. In this paper, we present and analyze various static and dynamic data allocation methods. The objective is to optimize the communication cost between a mobile computer and the stationary computer that stores the online database. Analysis is performed in two cost models. One is connection (or time) based, as in cellular telephones, where the user is charged per minute of connection. The other is message based, as in packet radio networks, where the user is charged per message. Our analysis addresses both the average case and the worst case for determining the best allocation method     

An ethnography of communication approach to mobile product testing

Product testing of mobile communication technology has typically employed the same research methodologies that were traditionally applied to stationary technology. An approach that does not primarily rely on physical location to study mobile communication technologies is thus needed. The stable component of mobile communication technology is not physical space but human communication. Therefore, a research model is developed based on an ethnography of communication approach, which designates “talk” (i.e., symbolic communication) as the primary and essential unit of measurement while making stationary physical location secondary. This allows design teams to enter a user “speech community” anywhere. Eight participants tested both the stationary and mobile version of customer relationship management software for sales. All participants were professional salespeople, comprising a speech community. Users articulated their “local” speech community meaning systems in the form of scenarios of use, which can guide product design and marketing. The findings show that proof-of-concept testing of mobile versions of desktop software can be done in conjunction with the usability testing for stationary technology.