基于多符号检测和Turbo乘积码的PCM/FM遥测系统性能

多符号检测(MSD)和Turbo乘积码(TPC)技术联合应用可以大幅提高脉冲编码调制/调频(PCM/FM)遥测系统性能。针对MSD算法计算复杂度高的问题提出了一种改进的MSD算法,可以有效降低计算复杂度;在TPC的传统Chase译码算法中通过简化软输入信息计算可以降低系统存储量。仿真结果表明,改进方法和传统的两种技术联合使用相比,虽然损失了约1.7dB解调增益,但仍提高PCM/FM信号解调性能约8dB,且算法复杂度低,存储量小,更适合硬件实现。     

浅谈通信系统中PCM常见故障处理

PCM设备是电力通信网中重要的接入设备,在电力调度通信自动化系统中起着重要的作用。PCM(Pulse Code Modulation,脉冲编码调制)终端复用设备是完成单路信号和多路信号(这里指2Mbit/s信号)的复接、分接设备,也是实现A/D转换不可缺少的设备,因其具有规范和灵活的接口、智能化程度高等特点,成为电力通信网的重要组成部分。本文着重介绍PCM的常见故障处理。     

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。     

基于改进的SOM网络模型的VoIP QoS应用研究

VoIP的服务质量（QoS,Quality of Service）评估可以采用一系列可度量的参数来描述：业务可用性、吞吐量、延迟、抖动、分组丢失率等。现有的感知语音质量评价（PESQ）很难对不同环境下的网络结构进行实时和恰当的语音等级质量分类。为了能够综合考虑几种QoS相关因素,在给出改进的自组织映射神经网络模型（ESOMNN）的基础上,利用ESOM能够对高维输入数据有效分类的特点,提出了将端到端延迟、丢包率、抖动、语音编码以及测试系统标识作为ESOMNN的输入数据,在对采样数据进行训练后可自动完成语音质量评价和映射,并能根据得到的实时变量有效地评价包含多种相关因素的QoS级别。     

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.     

Recognizing voice over IP: a robust front-end for speechrecognition on the world wide web

The Internet Protocol (IP) environment poses two relevant sources of distortion to the speech recognition problem: lossy speech coding and packet loss. In this paper, we propose a new front-end for speech recognition over IP networks. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bit stream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant benefits. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion due to the encoding-decoding process. Second, when packet loss occurs, our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec. We have considered the ITU G.723.1 standard codec, which is one of the most preponderant coding algorithms in voice over IP (VoIP) and compared the proposed front-end with the conventional approach in two automatic speech recognition (ASR) tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated packet loss rates. Furthermore, the improvement is higher as network conditions worsen     

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。     

基于声回波抵消两路算法的改进更新逻辑

声回波抵消两路算法被广泛用来检测系统双向通话;基于声回波抵消两路算法,提出了一种改进的控制更新逻辑。此更新逻辑通过比较滤波器的回波返回损失(ERLE),判断是否对滤波器进行更新。此改进更新逻辑能正确检测系统双向通话,避免滤波器的错误更新,并提高两路算法的收敛速度,减小存储器资源和计算量。仿真结果证实了此更新逻辑的有效性。     

骨干网中 VoIP语音质量的快速评估方法

提出一种基于被动模式流量分析的 VoIP语音质量评估方法 ,以帮助 VoIP运营商对主干网中大量并发 VoIP会话的通话质量进行实时监测。在流量采集的基础上 ,通过流跟踪算法获取 VoIP会话的丢包、时延等基本性能指标 ;通过快速评估模型对 VoIP会话的 R和 MOS值进行计算。实验证明 ,该方法性能较高 ,评估结果准确。     

VoIP over DVB-RCS satellite systems: Trial results and the impact of adaptive speech coding using cross-layer design

VoIP is one of the most popular applications on the Internet. The DVB-S2 (digital video broadcasting-satellite version 2) and DVB-RCS (digital video broadcasting-return channel satellite) standards provide the potential for reliable and efficient voice and data transfer over satellite networks. This is important, as satellites can play a role in broadband service provision by addressing the cases of limited terrestrial access such as in rural or underserved areas. The first part of this paper presents the results of our VoIP trials with different commercial DVB-S/RCS satellite offers and popular internet telephony and videoconferencing applications (Skype, MSN, etc.). These results reveal that packet delay and jitter are strongly affected by the satellite network component as well as the type of speech codec used. Accordingly, research presented in the second part of this paper is focused on dynamic speech coding rate control adapted to the conditions of the underlying network, in which the satellite domain presents the most challenging portion of the end-to-end path. For this purpose, a novel cross-layer mechanism is proposed to facilitate and increase the accuracy of the speech coding rate adaptation mechanism. Cross-layer design is a relatively new idea aiming to exploit information exchange among layers of the protocol stack. Our simulation analyses show that the proposed cross-layer mechanism can help us in optimally adjusting the speech coding rate to maintain the user-perceived quality in terms of MOS (mean opinion score) in the face of time-varying available satellite channel capacity.     

An unified VoIP model for workload generation

This paper presents a new model for VoIP workload generation. The novelty of our proposal consists in modeling the sessions by characterizing both the user behavior (session level) and the packet generation for an active call (intra-session level) with easily measured parameters and low computational complexity. This approach also facilitates systematic study of changes in user behavior and voice codec. The session level was modeled by analysis of call-holding time and time interval between successive calls. The model for call-holding time, characterizing the individual user behavior, uses the Pareto type 2 probability distribution. The time interval between calls is obtained from aggregate traffic and can be modeled by exponential probability distribution. Aggregate traffic is obtained by superposition of simultaneous sessions. The data used to characterize the session level were collected at the backbone of two Brazilian telecommunication carriers. The model for intra-session level comprises the characterization of the packet size and the packet inter-arrival time. The intra-session model was based on data generated in a laboratory environment, in order to properly characterize the codec influence on packet generation and to avoid the effects of delay, jitter and loss commonly present in an operational network. Models for constant bit rate and variable bit rate codecs were considered. A simulator was implemented and the results indicate that our model properly mimics the characteristics observed in real traffic and can be used for VoIP modeling and workload generation. Additionally, an application to automate the performance analysis was developed.     

Evaluation of objective speech transmission quality measurements in packet-based networks

This paper presents an analysis of the relation between IP channel characteristics and final voice transmission quality. The NISTNet emulator is used for adjusting the IP channel network. The transmission quality criterion is an MOS parameter investigated using the ITU-T P.862 PESQ, future P.863 POLQA and P.563 3SQM algorithms. Jitter and packet loss influence are investigated for the PCM codec and the Speex codec.     

二维空码索引调制算法

针对正交空间调制（QSM）提升传输速率时天线的使用数量增加,需要耗费大量资源且实现困难的问题,提出二维空码正交索引调制（SCOIM）。发射端信息比特分别映射为伪随机（PN）码的索引、天线的索引以及调制符号,调制符号的同相部分和正交部分再分别选择激活的PN码进行扩频,并各自通过激活的天线将信号发射出去。分析和仿真结果表明,相同传输速率时,SCOIM比正交空间调制节约至少一半的索引资源且随着传输速率的提升节约效果成倍增加,并且当误码率为10^-4时具备约5 dB的性能优势。     

Subjective MOS model and simplified E-model enhancement for Skype associated with packet loss effects: <Emphasis Type="Italic">a case using conversation-like tests with Thai users</Emphasis>

This paper proposes two mathematical models that can be used to estimate VoIP quality from Skype, which is one of the most popular VoIP applications. The first model is simple, it has been developed using data from the informal interview tests called Conversation-like tests, referring to packet loss of 0 %, 5 %, 10 %, …, and 30 %. The tests have been conducted with Skype using a non ITU-T’s codec called SILK via the Internet with over 180 native Thai participants, while packet loss effects were generated using a network emulation tool. The second model is called the Enhanced Simplified E-model, this has been developed by adding the Thai Bias factor into a generic Simplified E-model, which calculates by subtracting the subjective results from the computed results using the Simplified E-model formula. After obtaining the models, they were evaluated with the Test set from 36 native Thai participants (different from the other group of participants) using Mean Absolute Percentage Error technique (MAPE). It has been found that VoIP quality measurement performance of both models are classified as excellent and provide higher reliability and accuracy than the Simplified E-model. Subjective MOS model and Enhanced Simplified E-model error reduction compared to the simplified one was at about 21.9 % and 21.2 % respectively, which is the major contribution of this work.     

基于LV-BPL传输的Internet宽带接入系统研究

宽带低压电力线（LV-BPL）通信是一种重要的Internet接入技术,基于数字喷泉码与正交小波包编码调制提出了一种新的LV-BPL物理层系统模型。系统采用数字喷泉码作外码,块编码作内码,按块编码调制（BCM）映射到正交小波包调制时频平面。由于小波包稳定的正交性、自由的时频铺砌,数字喷泉码与码率无关的性质及BCM抗脉冲干扰的特点,系统在LV-BPL信道具有比编码DFT-OFDM更好的性能。     

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。     

基于四元数的双彩色图像盲水印算法

提出一种基于四元数的双彩色图像盲水印算法。对载体图像和水印图像分别进行离散四元数傅里叶变换后,对水印图像频域的低频、中频和高频系数做混沌置乱处理,并利用脉冲编码调制方法,在载体图像的低频、中频和高频嵌入置乱后水印的相应频带系数,通过逆离散四元数傅里叶变换得到嵌入水印的图像。实验结果表明,该算法保证水印具有较高的透明性,对剪切、叠加噪声、JPEG压缩等常见的攻击具有较好的鲁棒性。     

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.     

用于回波抵消的最大长度序列相关近端语音检测算法研究

回波抵消是语音通信系统中不可缺少的一个重要组成部分。大部分回波抵消技术都是基于自适应LMS算法的。在实际应用中，近端语音检测的准确性会在很大程度上影响自适应LMS算法在双向通信环境下回波抵消的效果。本文提出了一种基于最大长度序列相关算法的近端语音检测算法。这种近端语音检测算法和自适应LMS算法相结合，得到的回波抵消算法在模拟双向通信环境下的回波抵消效果比自适应LMS算法高约8db。     

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。