Profibus-DP总线单主站系统报文传输延时分析

采用确定与随机Petri网建立了Profibus-DP总线单主站系统报文传输延时的模型,并在TimeNet软件环境下对其进行了仿真分析,得出了报文传输延时与目标令牌循环时间的关系;并以STEP7软件为例,说明了在实际工程中如何根据Profibus-DP总线参数合理设定目标令牌循环时间、从而使监控系统获得最好的实时性的方法。仿真结果验证了该模型的正确性。     

无线传感器网络中带延时的一致性时间同步

针对无线传感器网络(WSNs)中节点间通信存在传输延迟,影响同步精度的情况,将加权平均应用于相对时钟斜率的计算,提出了一种带延时的一致性时间同步算法.该算法中每个传感器节点通过与邻居节点通信交换时钟信息,根据一致性理论更新时钟参数,从而到达时间同步的目的.研究了在假定传输延时服从正态分布的情况下对一致性时间同步算法的影响,提出的算法降低了延时对同步精度的影响,Matlab仿真验证了该算法的有效性.     

斜井人车话音与信号共信道传输的研究

根据话音通信的特点,介绍了一种提取实用的话音通断信号的方法,以及采用话音插空技术实现话音和简单数据共信道传输的复帧结构。     

网络混成传输话音接口板的设计

本文介绍话音接口板的工作原理与设计。该接口板的主要特点是将网络站点从模拟话音信号的处理中释放出来,从而实现话音与数据在网上的混成传输。本文最后介绍了提高话音质量的几个技术要点。     

一种基于帧内压缩的视频实时传输方法

利用通用的UDP传输实时视频,接收端所显示的视频图像往往存在延时较长、丢帧及单帧图像显示不完整等问题,这些现象在无线局域网环境中尤为明显.针对上述问题提出了一种基于帧内压缩的视频实时传输方法.在发送图像数据之前进行帧内压缩,然后分组发送;在接收端进行数据包接收、重组并解码.在局域网环境下进行了多次实验,对相关参数进行优化和调整,最终将延时减少到0.5～2 s.结果表明,由于采用了压缩算法并提高了拆包、组包的效率,视频传输的实时性得到了明显的改善,传输的准确性和稳定性都有明显的提高.     

图书馆多媒体信息同步传输管理方案的研究

文章研究了分步式多媒体信息系统同步传输管理方案,给出了传输过程中多媒体对象所必须遵循的时间约束基本要求,有界缓冲区大小的设置方法,延时抖动的估计方法,并在实际中得到了验证。为网络环境中多媒体对象同步传输进行了有益的探索。     

IP电话

IP电话是一种将分组话音数据经Internet传输的话音通信。本文介绍了IP电话的系统构成、工作原理、关键技术、通话质量等     

电视电话光缆—电缆混合传输系统光缆传输部分

已经研制成功的光缆——电缆信道互通电视电话传输系统,是由光信道机、话音转换控制器与改装的纵横制电话交换机构成,可同时进行图象、话音的转接,传输电视电话信息。电话拨号音、振铃、忙音等信号均能经光信道传输,经交换机与各信道互通。光信道机还可与没有电视电话端机的普通电话机通话。本文主要叙述光信道系统的研制简况。将调频的话音载频信号(2.0MHz)与基带视频信号(1MHz)复合放大后,对GaAlAS发光二极管(LED)进行直接强度调制,用阶跃型光缆作双工传输,接收端使用了低成本的Si-PIN光电二极管探测。接收机的灵敏度为-30.27dbm,未加权视频信噪比(Sp-p/Nrms)≥53db(调制度约50％),行时间非线性失真<2％,话音信噪比(Srms/Nrms)=42db,话音失真度小于1.5％(频偏7KHz)。使用1LED光源和3.5db/km损耗的阶跃型光纤,同时传输图象和话音,传输距离达3公里,话音清楚,直观图象清晰。     

视频流媒体的QoS特征分析及质量保证

网络数据传输的质量保证一般可分为空间域保证和时间域保证。空间域保证是传输数据的量的保证,取决于网络的带宽和丢失率特征;时间域的保证是对传输数据的时效性保证,取决于网络的延时和延时抖动特征。传统的数据媒体对网络传输的空间域保证有严格的要求,而没有严格的时间域保证要求。它不允许有数据丢失,但能够容忍一定的时间延时和延时抖动;而视频流媒体与此相反,它能够容忍一定的数据丢失,但要求较为严格的延时和延时抖动保证。这篇文章主要对视频流媒体数据传输的空间域特征和时间域特征进行讨论,分析其对视频流媒体的质量影响,为流媒体的网络传输QoS保证策略提供有价值的参考。     

基于信噪比的延时攻击防御方法研究

基于密码学的信息安全方法无法抵御延时攻击,提出一种基于信噪比的延时攻击防御方法。根据信噪比与传输功率的关系,综合考虑传输功率、信噪比和能耗的关系,建立合法传输节点和攻击节点的目标函数。使用博弈论方法,分析攻击节点,再分析合法传输节点。分析攻击节点与合法传输节点的传输信噪比、传输功率与能耗之间的权衡策略,得到合法传输节点传输功率的最优值。仿真实验分析表明,该策略提高了网络传输信噪比,减小了传输时延,防御了延时干扰攻击。     

Car Speech Enhancement Using a Microphone Array

This paper proposes a speech enhancement approach to suppress the interference of car noise. A linear microphone array is adopted for far-talking speech acquisition and delay-and-sum beamforming noise reduction. We present an effective time delay estimator using the coherence function between the reference microphone and the beamformed speech. To further enhance the beamformed speech, we exploit an improved Wiener filter where the resulting noise correlation in microphone array is relatively small so that the performance of optimal Wiener filtering could be achieved. Also, due to the serious degradation in low frequency car speech, we develop a spectral weighting function to compensate the low frequency filtering. These two processing units serve as the post filters to attain the desirable enhancement performance. In the experiments on microphone array speech in presence of real and simulated car noises, we find that the proposed algorithm performs well. Performance is measured in terms of the signal-to-noise ratio and the word error rate. The combined delay-and-sum beamformer and two post filters obtain the best results compared to other methods.     

Phase-Based Dual-Microphone Speech Enhancement Using A Prior Speech Model

This paper proposes a phase-based dual-microphone speech enhancement technique that utilizes a prior speech model. Recently, it has been shown that phase-based dual-microphone filters can result in significant noise reduction in low signal-to-noise ratio [(SNR) less than 10 dB] conditions and negligible distortion at high SNRs (greater than 10 dB), as long as a correct filter parameter is chosen at each SNR. While prior work utilizes a constant parameter for all SNRs, we present an SNR-adaptive filter parameter estimation algorithm that maximizes the likelihood of the enhanced speech features based on a prior speech model. Experimental results using the CARVUI database show significant speech recognition accuracy rate improvement over alternative techniques in low SNR situations (e.g., an improvement of 11% in word error rate (WER) over postfiltering and 23% over delay-and-sum beamforming at 0 dB) and negligible distortion at high SNRs. The proposed adaptive approach also significantly outperforms the original phase-based filter with a constant parameter. Furthermore, it improves the filter's robustness when there are errors in time delay estimation     

基于不同长度拼接单元的英文文语转换系统

提出用不同长度的单元进行拼接的英语语音合成方法。实验表明,该方法能更好地利用自然语流的原始信息,提高合成语音的自然度,同时也能提高系统的灵活性和鲁棒性。     

利用PCNN神经元点火位置信息的语音特征研究

语谱图是语音信号的时频表示,含有丰富的信息。把语谱图输入到脉冲耦合神经网络（PCNN）可以获得语音的特征矢量。传统的语音特征采用PCNN50次迭代的点火次数。提出了一种新的语音特征参数,该参数基于PCNN神经元点火位置的信息。说话人识别的实验表明,新语音特征比传统的特征更能反映话者语音信号的特点,获得更好的识别结果。     

基于相空间重构的语音特征研究

本文通过重构语音信号相空间。研究语音的相似序列重复度及其熵信息,分析比较了语音信号在相空间中的非线性特征。根据清音和浊音在多维相空间中的不同空间分布特性,对语音音素进行了分类。利用语音信号在相空间中的非线性特征可以为语音识别研究提供一个新的方向。     

TMS320C54X实现ITU G.728语音编码标准

ＩＴＵ Ｇ．７２８标准晃国际电信联盟于１９９２年的一种比４特率为１６Ｋｂ／ｓ低延时ＣＥＬＰ类型语音编解码器。本文首先对Ｇ．７２８编解码算法和定点数字处理芯片ＴＭＳ３２０Ｃ５４Ｘ作了简单介绍。由于Ｇ．７２８标准是一种低延时语音编码标准,因此计算复杂度高,在实时实现中需要作特别处理。本文着重介绍了这种低延时ＣＥＬＰ（ＬＤ－ＣＥＬＰ）算法在ＴＭＳ３２０Ｃ５４Ｘ上实现的软、硬件设计和在定点ＤＳＰ芯片实现此     

基于LTE系统的VoIP自适应调度算法

提出一种基于LTE系统的VoIP服务的自适应上行调度算法,该算法采用自适应多速率语音编码器,利用传统MAC通用报头中的2个比特将语音编码的模式告知eNB,eNB根据UE的语音状态转换和语音编码速率动态分配上行链路资源。从系统容量、吞吐量和时延方面对比分析该算法和传统算法的性能。理论分析和仿真结果表明,在时延满足要求的前提下,该算法比传统算法具有更高的系统容量和吞吐量。     

Packet loss concealment method based on interpolation in packet voice coding

This paper proposes a packet loss concealment (PLC) method based on redundant information to enhance speech quality under severe network conditions as well as a new model to simulate packet loss effect, end-to-end transmission delay, and jitter on the concealment process. Our proposed method increases the efficiency of the standard ITU-T G.722.2 by improving the quality of the decoded speech under random frame loss conditions over packet-switched networks. The new method achieves a MUltiple Stimuli with Hidden Reference and Anchor (MUSHRA) value exceeding 48.7 at a 20% packet loss ratio (PLR). Numerical analysis indicates that our method outperforms existing alternatives and can be successfully used, even in severe propagation scenarios at the expense of an increase in processing requirements.     

The combined effect of speech codec quality and transmission delay on human performance during complex spoken interactions

This paper examines the effect of interaction between speech codec output quality and simulated satellite or VoIP transmission delay time on talker performance in a complex interaction. A hardware test codec (both single and tandem) was compared against a number of processed speech reference conditions to determine the relative subjective quality of the test codecs against conditions with known Mean Opinion Scores (MOS). The two codec conditions plus an additional higher quality condition were then used in an experiment that examined the effect of the interaction of transmitted speech quality and simulated transmission delay on a speech shadowing task and an accompanying error repair task involving two speakers. One person (the “reader”) read a passage. The second person (the “shadower”) shadowed the read passage by repeating immediately the words spoken by the reader. The reader, whilst reading, also listened for errors spoken by the shadower and repaired those errors by verbally reporting them to the shadower. A significant interaction between codec quality and transmission delay was found for the error repair task, but only for cases where the shadower made a significant number of errors. These results suggest that, for highly complex interactions which involve significant cognitive load, human performance will degrade more rapidly with increases in delay for transmission systems using speech codecs with lower quality output. This is assumed to be due to the additional demands upon working memory imposed by the transmission delay.     

Analog scramblers for speech privacy

Despite significant strides in digital speech technology, scrambling of the analog speech signal continues to be an important method for achieving casual privacy in many classes of voice communications systems. This paper explains the principles of important classes of analog speech scramblers, with emphasis on algorithms that possess demonstrated robustness in the context of real-channel operation. At the forefront of the art are so-called two-dimensional analog scramblers that manipulate the speech signal in both time and frequency domains, to obtain impressive combinations of residual intelligibility, communication delay and transmission robustness. Our review includes several examples of such two-dimensional algorithms.