数字会议网络系统

数字会议网络（ＤＣＮ）具有讨论系统、同声传译系统等综合性功能，适用于从小型研讨会议到大型国际会议的不同要求。文章主要介绍了ＤＣＮ的基本组成方式，重点论述了同声传译系统的构成，最后介绍了一个工程应用实例。     

数字会议和同声传译系统发展及应用

１会议和同声传译系统简介会议和同声传译系统是举行会议时使用的用于会议管理的电子系统。会议系统在国外有时被称做讨论系统。通常这样的系统是单一语种系统，而同声传译系统则是多语种的系统，一般要和会议系统相结合。随着国际交流和合作的日益频繁，国际性的会议越来越多，来自不同国家和地区的代表用自己熟悉的语言进行发言讨论，这就需要有一套同声传译系统将发言的内容翻译成几种使与会代表都能听懂的语言。目前，同声传译系统已成为国际性会议厅的必备设施。２会议和同声传译系统的现状２．１国外国外的会议和同声传译系统主要生产…     

TAIDEN同传系统完美服务于2008年北京重大赛事

2008年8月8日和9月6日在北京举行了体育重大赛事,所有重要场馆均采用了由深圳市台电实业有限公司提供的会议系统、同声传译系统和相关服务。此次中标使用的“TAIDEN”会议系统和同声传译系统的服务场所,包括新闻发布厅及国家体育场、国家游泳中心、国家体育馆、五棵松篮球馆等场馆的新闻发布厅,实现了会议发言、讨论、同声传译、录音录像等功能。其中,国家体育场新闻发布厅使用了深圳台电公司最新研制的高端会议同传产品TAIDEN HCS-5100数字红外语言分配系统。     

基于云桌面技术的高校同声传译语音室远程控制方法

为了提升远程通信语音质量及数据传输、访问的时效性，研究基于云桌面技术的高校同声传译语音室远程控制方法。分析并构建云桌面技术的主要功能结构，应用此功能结构结合Phantosys DVP架构，设计Phantosys DVP虚拟云桌面的远程控制实施方案，实现对高校同声传译语音室的远程控制，在此基础上，运用回声消除算法针对远程控制过程中产生的远程通信语音信息内回声实施处理，提升远程通信语音信息的质量，实验结果表明，该方法的E_RLE值较高，远程控制过程中数据丢包率较低，能够保证数据传输的完整性与时效性。     

同声传译系统原理及应用

同声传译系统在使用不同语言的会议场合 ,发挥着重要的作用。在北京人民大会堂里就装有八种语言 (在每年的“两会”中经常使用汉、维吾尔、哈萨克、蒙、藏、壮、彝、朝鲜等语言 )的同声传译设备 ,以便使参加会议的各民族代表能同步地聆听大会的发言。同传系统更为国际会议带来更高的工作效率。我国改革开放以来 ,与国外经济、技术、政治、文化的交流日益增多 ,国际研讨会也日益频繁 ,使用同传设备的机会越来越多。本文仅就目前经常使用的同声传译系统原理及应用介绍如下。1同声传译系统组成同声传译系统是将发言者的语言 (通常称原语— Ori…     

会议系统行业新纪元——国家标准《红外线同声传译系统工程技术规范》发布

对GB50524—2010的主要内容及编制过程进行了介绍，同时介绍了红外线同声传译系统，阐述了BANDII频段的红外同传系统淘汰的原因，说明了BANDIV频段的红外线同声传译系统的重要性及对会议系统行业带来的影响。     

用调音台实现同声传译的研究

１引言随着２１世纪的到来，特别是中国加入WTO进程的加快，国际交流与合作将愈加频繁，国际性的会议越来越多。来自不同国家和地区的代表在发言、讨论时使用的语种是不同的，在这种多语种的环境下就需要同时把发言者的语言翻译成多种其他听众的语言，这时就需要同声传译会议系统，又称同声翻译会议系统。同声传译会议系统已成为国际性会议厅的必备设施。目前，生产同声传译系统的厂家很多，但绝大多数都是国外的厂家，产品品质好、功能全，但价位很高，对于一些临时性会议，再加上场地的限制，安装、调试也很不方便，为此笔者研究设计了…     

浅析同声传译中的同步笔译

同声传译是目前国际大型会议常用的一种口译模式,同声传译也被视为口译者的最高境界。对于如何练成同声传译,或者同声传译练习有哪些技巧,则是智者见智仁者见仁。与很多将同声传译与笔译进行对比得出相应的练习方法不同,笔者从同声传译自身的特点出发,提出了“同步笔译”这一概念。以期能够对同传教学和同传练习提供借鉴和参考。     

联创：开拓渠道 推广品牌

广东省联创科尔电声器材厂以研发和生产中高端会议系统为主,随着市场需求的不断提高,逐步推出了数字表决会议系统、无线表决系统、IC卡签到系统、视像跟踪系统、同声传译会议系统等产品。在Bekrl品牌实现在国际市场成功销售后,近几年开始以MCMYK品牌进入国内市场。     

公信DSSS无线数字同传系统正式取得专利证书

由佛山市公信数字会议设备有限公司开发的“DSSS无线数字同声传译系统”日前正式取得了国家知识产权局颁发的2个实用新型专利证书。2个专利证书名称分别为：（1）实用新型专利：8位单片机与16位串口音频编码解码器连接的接口装置。     

An analysis of the performance of ARQ schemes for integrated traffic satellite-switched SFH and DS CDMA networks

The delay and throughput performance of satellite-switched Slow Frequency Hopping CDMA network for simultaneous voice and data transmission is analyzed and compared to that of a DS-CDMA system. Two ARQ schemes are suggested for data while Forward Error Correction using the same encoder is used for voice packets. The queueing analysis assumes priority for voice and two models for voice traffic are used (Markovian and IPP). The probability of successful packet transmission is derived for all systems as a function of traffic load allowing us to evaluate the systems using delay, throughput, and voice packet loss as figures of merit. Numerical results show that while voice delay is minimal DS CDMA is much more effective then SFH CDMA in all cases. One interesting result is that SFH systems perform better with S/W schemes and achieve a higher maximum throughput. It is also observed that the IPP and Markovian models gave similar results.This work was supported by an NSERC CRD (Collaborative Industrial Research and Development grant,) with Spar Aerospace, Quebec, Canada     

Simultaneous voice and data on the general switched telephonenetwork using framed QADM

Considerable effort has been expended to allow voice to be transmitted over a general switched telephone network (GSTN) connection simultaneous with modulated data. A new transmission technology for this purpose is introduced-framed quadrature audio/data modulation (framed QADM)-in which voice and data channels are combined as an integral part of the modulation process. The basic technology is described and two implementations, which are at different levels of maturity for standardization within the International Telecommunications Union (ITU), are presented. Finally, system attributes are discussed for framed QADM as they compare to other technologies     

Self-subrating for voice/data in PCS

Bellcore Personal Access Communications System (PACS) provides the feature of using a single time-slot for two independent calls by the same user. This feature allows a simultaneous voice and data call from a single subscriber unit. This paper compares the performance of a PCS system with voice/data self-subrating (SSR) with a system without self-subrating (NSSR). We show that for the ranges of the input parameters we study, the blocking probability for NSSR is 76% higher than for SSR. For a PCS system engineered at 1% blocking probability, SSR carries 15.3% more offered load than NSSR.     

Adaptive hierarchical modulation for simultaneous voice and multiclass data transmission over fading channels

In this paper, a new technique for simultaneous voice and multiclass data transmission over fading channels using adaptive hierarchical modulation is proposed. According to the link quality, the proposed scheme changes the constellation size as well as the priority parameters of the hierarchical signal constellations and assigns available subchannels (i.e., different bit positions) to different kinds of bits. Specifically, for very bad channel conditions, it only transmits voice with binary phase-shift keying (BPSK). As the channel condition improves, a variable-rate adaptive hierarchical M-ary quadrature amplitude modulation (M-QAM) is used to increase the data throughput. The voice bits are always transmitted in the lowest priority subchannel (i.e., the least significant bit (LSB) position) of the quadrature (Q) channel of the hierarchical M-QAM. The remaining (log/sub 2/M-1) subchannels, called data subchannels, are assigned to two different classes of data according to the selected priority parameters. Closed-form expressions as well as numerical results for outage probability, achievable spectral efficiency, and average bit error rate (BER) for voice and data transmission over Nakagami-m fading channels are presented. The adaptive techniques employing hybrid binary shift keying (BPSK)/M-ary AM (M-AM) and uniform M-QAM for simultaneous voice and two different classes of data transmission are also extended. Compared to the extended schemes, the new proposed scheme is spectrally more efficient for data transmission, while keeping the same outage probability for voice and data (both classes) as the scheme employing BPSK/M-AM. The new scheme also provides, as a by-product, a spectrally efficient way of transmitting voice and a single-class data.     

Approximate performance analysis of heavily loaded slotted downlink channel in a wireless CDMA system supporting integrated voice/data services

This paper is concerned with the performance analysis of a slotted downlink channel in a wireless code division multiple access (CDMA) communication system with integrated packet voice/data transmission. The system model consists of a base station (BS) and mobile terminals (MT), each of which is able to receive voice and/or data packets. Packets of accepted voice calls are transmitted immediately while accepted multipacket data messages are initially buffered in first in, first out (FIFO) queues created separately for each destination. The BS distinguishes between silence and talkspurt periods of voice sources, so that packets of accepted data messages can use their own codes for transmission during silent time slots. To fulfill QoS requirements for both traffic types, the number of simultaneous packet transmissions over the downlink channel must be limited. To perform this task, a fair, single-priority multiqueueing scheduling scheme is employed. Discrete-time Markov processes are used to model the system operation. Statistical dependence between queues is the main difficulty which arises during the analysis. This dependence leads to serious computational complexity. The aim of this paper is to present an approximate analytical method which enables one to evaluate system performance despite the dependence. Therefore, it is assumed that the system is heavily loaded with data traffic, and a heuristic assumption is made that makes the queueing analysis computationally tractable. Typical system performance measures (i.e., the data message blocking probability, the average data throughput and delay) are evaluated, however, due to the accepted heuristic assumption, the analysis is approximate and that is why computer simulation is used to validate it.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

基于MATLAB的皮肤听声器系统的研究

为了使皮肤听声器能够辨析语音,设计了基于MATLAB的皮肤听声器系统。该系统以MATLAB软件为平台,在此基础上实现了语音信号的录制、播放、预处理、分段滤波、特征提取等功能,并利用特征参数辨析语音。本系统达到了辨析简单语音的要求,但仍有需改进的地方,如:能否构造更合适的特征参数,能否辨析比较复杂的语音。     

基于以太网的数字话音传输系统设计

针对在工程应用中如何通过以太网进行话音传输提出了一种设计方案,分析了话音在网络上传输的特点,介绍了一种基于以太网的数字话音传输系统方案。系统以自带网络协议的嵌入式ARM微控制器LM3S9B96为核心平台,采用IP上传送语音（Voice over IP,VoIP）技术实现话音的以太网传输。对系统的话音实际传输效果进行了仿真测试分析,结果表明,话音清晰、失真度和时延小,整体性能满足实际话音通信的要求。     

Adaptive subcarrier allocation and bit loading for voice/data transmission in multiuser OFDM systems

A new algorithm of adaptive subcarrier allocation and bit loading (A‐SABL) is proposed for simultaneous voice and data transmission in multiuser OFDM systems. The algorithm takes advantage of the frequency diversity and the voice/data transmission requirements to dynamically assign the number of subcarriers and bits/per symbol on each subcarrier for each user in a single cell. Due to the strict delay requirement of voice service, the subcarriers with low channel gains are assigned for voice transmission with a small number of bits per symbol to guarantee its required bit‐error‐rate (BER) and transmission rate. Based on the remaining subcarriers with high channel gains and the transmission power, the throughput of data transmission is then maximized by loading as many bits as possible on each subcarrier to achieve the required transmission bit rate and BER. Theoretical analysis and simulation on the proposed algorithm show that a better performance is obtained than previously reported schemes. Copyright © 2008 John Wiley & Sons, Ltd.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls