基于语音质量预测的Speex自适应码率控制算法

为实现IP语音(VoIP)质量的动态管理与控制,提出了一种基于语音质量预测的自适应码率控制算法。通过实时预测VoIP通话的瞬时语音质量和总体语音质量,自适应地调整Speex编码参数,从而根据需要选择最佳编码速率。实验仿真结果表明,提出的算法能够有效减少网络拥塞,提高VoIP系统的语音质量。     

基于E-Model的VoIP语音质量研究

针对目前网络电话语音质量难以准确评价及预测的情况,基于E-Model对VoIP的语音质量进行预测。分析几个主要影响因素,如延时、丢包等对话音质量的影响,构建VoIP语音质量预测模型,将E-Model中未考虑到的抖动因素引入模型公式,着重考虑抖动缓冲区的大小对语音质量的影响。通过设计相关验证实验,证明该模型对VoIP语音质量的预测具有较高的准确度。     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

一种新的VoIP自适应缓冲算法

网络延迟与缓冲的矛盾是VoIP应用中的一个重要问题.介绍了VoIP应用中几种当前主要的缓冲算法,分别分析了它们的优缺点,提出了新的自适应缓冲算法,称为FISD算法,对现有的代表性算法以及FISD算法分别进行了仿真实验.结果表明,在网络延迟抖动较大时,新算法可以有效地提高语音质量.     

一种自适应IP语音缓冲算法的研究与应用

当IP语音包的网络时延抖动较小时,一般的语音缓冲算法可以得到较好的语音质量．当网络中存在突发大时延时,就会出现极大丢包率或极大端到端时延,从而难以获得好的语音质量．为此,提出针对突发大时延下的自适应语音缓冲算法．通过估算网络平均时延和学习语音包经过的网络路径上的状态,来确定需要控制端到端时延大小和语音包的丢包率,动态调整Jitter Buffer队列的最小深度和最大深度,从而可以尽量减小语音裂缝（gap）的出现．通过基于听觉模型的客观音质评价（PESQ）仿真计算以及在实际语音网关设备中的应用表明算法对语音通信质量有一定的改善作用．     

Vo IP 的语音质量分析与控制

分析了VoIP语音质量的影响因素，通过E模型定量地描述了语音质量与端到端延迟和丢包率的关系。为了控制VoIP的语音质量，计算出VolP系统在各种情况下的语音质量极限，提出一种自适应编码和分组封装的控制策略。将该方法应用于自行开发的IP电话网关，实际测试证明能在很大程度上提高VoIP的语音质量。     

VolP的关键技术

结合VoIP的硬件、软件、协议和标准,从语音压缩编码技术,IP网的延时和抖动的处理技术、回音消除和静音压缩技术等几方面展开了研究和探讨,并展望了VoIP的发展前景。     

基于时间尺度调整技术的自适应缓冲新算法

介绍了VoIP应用中几种主要的动态缓冲算法,分析了各自的实现方式和性能优劣,提 出了基于简化的相似波形叠加技术和双窗预测技术的自适应缓冲算法,并进行了仿真。结果表明,新 算法能明显提高IP电话的语音服务质量。     

基于节点网络深度的缓冲延时算法研究

基于ZigBee技术的语音导游系统,语音数据包在游客接收端会产生失序问题。为了消除这种语音抖动,提出了一种基于游客节点网络深度的缓冲延时算法,采用E-Model语音预测模型,用客观的语音预测值去表示主观的MOS（Mean Opinion Score,平均意见值）评分值,通过计算最高MOS值得到对应的最优延时变量。通过简化算法并软件仿真,可以看出该算法在语音导游系统中的优越性。     

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。     

基于E-model的VoIP语音质量评估的研究

为准确评估VoIP）语音质量，对E—model算法进行了深入研究，剖析了E—model算法的组成部分Id，Is，Ie，A，探讨了丢包、延迟和抖动对VoIP质量的影响，并应用该算法对多种语音编码进行了评估。实验证明该客观评估算法主观与客观相关度高，有较强的适应性，可靠性，实用性，完全可用于VoIP语音质量评估。     

A Garch-based adaptive playout delay algorithm for VoIP

Network delay, packet loss and network delay variability (jitter) are important factors that impact on perceived voice quality in VoIP networks. An adaptive playout buffer is used in a VoIP terminal to overcome jitter. Such a buffer-control must operate a trade-off between the buffer-induced delay and any additional packet loss rate. In this paper, a Garch-based adaptive playout algorithm is proposed which is capable of operating in both inter-talkspurt and intra-talkspurt modes. The proposed new model is based on a Garch model approach; an ARMA model is used to model changes in the mean and the variance. In addition, a parameter estimation procedure is proposed, termed Direct Garch whose cost function is designed to implement a desired packet loss rate whilst minimising the probability of consecutive packet losses occurring. Simulations were carried out to evaluate the performance of the proposed algorithm using recorded VoIP traces. The main result is as follows; given a target Packet Loss Rate (PLR) the Direct Garch algorithm produces parameter estimates which result in a PLR closer than other algorithms. In addition, the proposed Direct Garch algorithm offers the best trade-off between additional buffering delay and Packet Loss Rate (PLR) compared with other traditional algorithms.     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

基于网络流量预测的抖动缓冲控制算法

根据网络流量的变化规律,提出自回归模型下的抖动缓冲控制算法。通过改进的随机中点置位算法,建立具有突发性和自相似 性的网络业务流量预测模型。依据网络流量的预测值设置缓冲区大小,并在使用中不断改进以提高缓冲区设置精度。采用Matlab软件对抖动缓冲控制算法进行仿真,基于E-modle对语音质量进行评估,结果表明该算法在没有其他服务质量保证的情况下,MOS值均在中级标准以上,较好地改善了网络电话的语音质量。     

基于抖动激励的VoIP终端拥塞控制机制

针对基于H．323的VoIP系统，提出了一种抖动激励式终端拥塞控制机制，将语音分组的延迟抖动参数与RTCP协议相结合，控制动态改变语音编码方式及传输分组大小，从而缓解网络拥塞，降低传输延时和丢包率。     

Adaptive VoIP playout scheduling: assessing user satisfaction

Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics.     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.