P.563—The ITU-T Standard for Single-Ended Speech Quality Assessment

Objective voice quality assessment has been the subject of research for many years. Up until very recently, objective models required a copy of the unprocessed signal for estimating the quality of a signal transmitted across a telecommunication network, making live call monitoring impossible. This paper introduces a method for nonintrusive assessment of speech quality for narrow-band telephony, which was approved by the International Telecommunication Union (ITU-T) in May 2004. Essentially based on models of voice production and perception, the algorithm demonstrates good performance on more than 48 subjective experiments representing most distortions that occur on voice networks.     

专用语音信箱数据采集与处理

介绍的专用语音信箱由２ 条中继线、７ 台分机、１ 个语音处理单元以及呼叫处理程序和语音信箱管理程序组成。内外线电话能够互相呼叫或拨号访问信箱,并拥有多种程控业务新功能。语音信箱具有查询、留言和播放公众信息等功能,在被叫忙音或无应答时自动进入信箱。呼叫处理程序使用状态迁移法解决呼叫信号采集、处理的多重性问题;使用时间调度技术解决多用户的实时处理问题;使用ＶｉｓｕａｌＢａｓｉｃ的ＭＳＣＯＭＭ 控件实现了串行口交互通信;将多媒体ＭＩＣ控件用于语音信息的记录、储存和重放过程。该语音信箱有明显的实用价值     

基于Dialogic语音卡实时数据采集的电话语音识别系统

语音识别技术在新一代呼叫中心的IVR系统中得到了广泛的应用。为了使用Dialogic电话语音卡进行语音识别,文章解决了用Dialogic语音卡进行语音数据实时采集的问题,并给出了一种用动态背景噪声电平检测语音的算法,建立起了基于DialogicD/120JCT-LS电话语音卡的自动电话交换转接系统。     

CDP技术在话音漫游清算容灾系统中的应用

随着话音漫游业务的快速发展,话音漫游清算涉及的领域不断扩大,在重大事故,自然灾害等突发事件发生时确保清算数据的可用性和业务连续性就显得尤为重要.针对话音漫游清算系统的容灾需求以及有待改进的问题,通过研究分析CDP(Continuous Data Protection)容灾技术,提出将CDP技术运用到话音漫游清算容灾系统的方法,设计并构建了一套具有持续数据保护功能和全面数据恢复能力的容灾系统.实践证明,该方法能够在各种灾难场景中有效地保障话音漫游清算系统的数据完整性和业务连续性.     

On-line experimental methods to evaluate text-to-speech (TTS) synthesis: effects of voice gender and signal quality on intelligibility,naturalness and preference

Three experiments are reported that use new experimental methods for the evaluation of text-to-speech (TTS) synthesis from the user's perspective. Experiment 1, using sentence stimuli, and Experiment 2, using discrete “call centre” word stimuli, investigated the effect of voice gender and signal quality on the intelligibility of three concatenative TTS synthesis systems. Accuracy and search time were recorded as on-line, implicit indices of intelligibility during phoneme detection tasks. It was found that both voice gender and noise affect intelligibility. Results also indicate interactions of voice gender, signal quality, and TTS synthesis system on accuracy and search time. In Experiment 3 the method of paired comparisons was used to yield ranks of naturalness and preference. As hypothesized, preference and naturalness ranks were influenced by TTS system, signal quality and voice, in isolation and in combination. The pattern of results across the four dependent variables – accuracy, search time, naturalness, preference – was consistent. Natural speech surpassed synthetic speech, and TTS system C elicited relatively high scores across all measures. Intelligibility, judged naturalness and preference are modulated by several factors and there is a need to tailor systems to particular commercial applications and environmental conditions.     

Silog: Speech input logon

Silog is a biometric authentication system that extends the conventional PC logon process using voice verification. Users enter their ID and password using a conventional Windows logon procedure but then the biometric authentication stage makes a voice over IP (VoIP) call to a VoiceXML (VXML) server. User interaction with this speech-enabled component then allows the user’s voice characteristics to be extracted as part of a simple user/system spoken dialogue. If the captured voice characteristics match those of a previously registered voice profile, then network access is granted. If no match is possible, then a potential unauthorised system access has been detected and the logon process is aborted.     

基于Bessel函数展开的ICA语音增强

将源信号的先验知识以参考信号的形式引入到独立分量分析(ICA)学习算法中,从混合信号中仅提取期望的源信号。依据语音信号传播机理和Bessel函数展开系数对语音信号的表征能力,给出基于Bessel函数展开的参考信号构建方法,从混合语音信号中提取出期望的语音信号。仿真和性能分析结果表明,该方法能在噪声干扰的情况下达到语音增强的目的。     

Exploiting hierarchical terrestrial-satellite architectures to handle voice,symmetric data,and asymmetric data connections

This article investigates new techniques for the dimensioning and the effective exploitation of radio resources in integrated terrestrial-cellular and satellite systems. A dynamically configurable radio resource management policy is proposed. Its main feature is the differentiated handling of user mobile calls, based on the type of service (voice, symmetric data, asymmetric data), the nature of the call (originating or handoff call) and the user mobility class (pedestrian or vehicular). The output of the study is particularly useful to support the design of next-generation wireless systems, which are based on a hierarchical cell structure where different cell types (e.g., macrocell and microcell) operate together with an overlapping satellite coverage.     

中、低速率话音编码技术在卫星通信中的应用

本文根据卫星通信对话音编码的要求,介绍了几种主要的中、低速率话音编码技术的现状与发展。并对这几种话音编码方式的性能进行了对比分析。探讨了在卫星通信中使用这几种话音编码方式的必要性和可行性。最后提出了卫星通信应用中、低速率话音编码方式的建议。     

基于语音拖音段的端点检测算法研究

为了进行有效的语音信号处理,并降低语音信号的冗余度,通常采用端点检测技术来提取语音信号中的有效部分。本文在传统语音端点检测方法的基础上,提出了一种基于基音周期对语音段末尾进行判别的方法,针对汉语发音都是以浊音结尾的特点,同时利用基音周期对浊音段信号比较敏感这一特性,能够有效地避开汉语语音信号尾部拖音段中所包含的无效信息,既提高了端点检测的准确性,又减少了后续语音识别系统样本训练时间。实验结果证明,该方法对于汉语中孤立词末尾的拖音段,可以得到较好的端点检测效果。     

基于翻转梅尔频率倒谱系数的语音变调检测方法

语音变调常用于掩盖说话人身份,各种变声软件的出现使得说话人身份伪装变得更加容易。针对现有变调语音检测方法无法判断语音是经过了何种变调操作（升调或降调）的问题,通过分析语音变调在信号频谱,尤其是高频区域留下的痕迹,提出了基于翻转梅尔倒谱系数（IMFCC）统计矩特征的电子变调语音检测方法。首先,提取各语音帧IMFCC及其一阶差分;然后,计算其统计均值;最后,在该统计特征上利用支持向量机（SVM）多分类器的设计来区分原始语音、升调语音和降调语音。在TIMIT和NIST语音集上的实验结果表明,所提方法无论对于原始语音、升调语音还是降调语音都具有良好的检测性能。与MFCC作为特征构造的基线系统相比,所设计的特征的方法明显提高了变调操作的识别率。在较少的训练资源的情况下,所提方法也获得了比基于卷积神经网络（CNN）的框架更好的性能;此外,在不同数据集和不同变调方法上也都取得了较好的泛化性能。     

基于改进谐波恢复算法的语音增强方法

当信噪比较低时,语音信号的高次谐波部分会完全淹没在噪音中。针对该情况,提出一种基于改进谐波恢复算法的语音增强方法。对经过MMSE-LSA算法语音增强处理后的时域输出语音信号进行非线性处理,得到准周期冲激信号,并将其与原增强信号相乘,突出语音的谐波分量。实验结果表明,改进算法较好地解决了低信噪比时谐波失真的问题,相比传统谐波恢复算法能更好地改善语音高次谐波的质量。     

基于分形维数和模糊RBF神经网络的语音端点检测

简单介绍了分形维数的概念及模糊RBF神经网络的结构。利用分形维数在噪声情况下作为语音端点检测参数的优越性,组合幅度熵、帧能量及过零率作为模糊神经网络的输入参数进行语音信号端点检测。用连续语音进行非正式测试,实验证明该方法避免了选取阈值这一难点,在噪声情况下仍具有较高检测准确率。     

卫星通信系统“动中通”与“静中通”技术在指挥应急现场工作中的应用

在指挥应急现场工作中,应急通信保障任务非常重要。数据、话音、实时监控信息等信号需要及时、准确地送到指定场所。在一些特定场合就需要利用卫星通信车来传送数据、话音、实时图像监控信息等信号。本文对卫星通信系统“动中通”和“静中通”技术进行研究,并对这两种卫星车的应用特点、应用场合做比较。     

Enhancing VoIP service for ubiquitous communication in a campus WLAN with partial coverage

The IEEE 802.11 WLAN technology has become the de facto standard for wireless Internet access. The spotty coverage of WLAN access points, however, confines the applicability of many real-time services such as VoIP within the boundary of the WLAN service area. In this paper, we investigate the problem of enhancing VoIP service for ubiquitous communication in a WLAN with spotty service area. We consider a university campus that has an established infrastructure for supporting SIP-based VoIP service through either wired or wireless data networks. The campus WLAN service does not have 100% full coverage, and hence users cannot make untethered VoIP calls anywhere on campus. The goal of this paper is to overcome the limitations of such “dead spots” for motivating the use of campus IP telephony service. To proceed, we start with two approaches called one-hop extension and dual-mode communication. The first approach uses multi-hop relay to extend the WLAN coverage, while the second approach leverages the availability of dual-mode handsets for ubiquitous voice communication. We implement the two approaches, and evaluate their performance in the campus testbed environment. We find that while the two approaches can effectively allow voice communication in WLAN dead spots, they have one common problem as the potential lack of support for voice call continuity that can cause degradation of the speech quality to an active call. We adopt a cross-layer solution based on signal processing algorithms to address the problem, thus achieving seamless voice call continuity while enabling ubiquitous voice communication on campus. Testbed evaluation shows promising results for future research along the proposed direction.     

保护信道和排队相结合的信道分配策略

提出了一种适用于语音、数据呼叫的蜂窝移动通信系统的信道分配策略。该策略为数据呼叫提供保护信道,降低数据呼叫的阻塞率。同时,采取语音呼叫排队策略抑制数据保护信道引起的语音呼叫阻塞率的恶化。为了进一步提高系统的性能,在策略中引入了不耐烦顾客,并建立了带有不耐烦顾客的排队模型。仿真结果表明该策略能够有效地降低语音呼叫和数据呼叫阻塞率,改善系统性能。     

语音频谱分析仪设计

语音频谱分析已经广泛应用到医疗、语音教学、语音编码、语音增强和语音识别等领域,并在这些领域发挥着非常重要的作用。这里介绍了语音信号频谱分析的原理及智能语音信号频谱分析仪的硬件结构和软件设计。语音信号频谱分析仪由32位ARM为主控制器,通过ARM自带的A/D转换接口,对语音信号进行采样,并以数组形式保存,然后通过FFT快速傅氏变换运算得到频谱,再通过高分辨率的液晶屏对语音信号的频谱进行实时显示。     

基于Asterisk平台的IP_PBX设计

基于IP方式的语音交换系统,是将基于IP协议的分组交换网用来传递电话语音信号,通过SIP、RTP/RTCP等协议来控制传输过程,有效的改善通话质量,解决存储转发方式用于语音信号传输存在的缺陷。     

Cross-correlation portraits of voice signals in the problem of recognizing voice commands according to patterns

A way for recognizing voice commands (VCs) in the noises with a probability of proper recognition higher than 92% and a signal/noise ratio of 1–6 dB, if the library of pattern voice commands has been generated directly before recognition, is presented in [1]. This method is based on transformation of voice signals into a 2D image: autocorrelation portrait (ACP). The results become significantly worse if the library is prepared long before the recognition, and this is a disadvantage of this method. In this paper we describe the procedure for generating another type of voice command image, which eliminates (to a considerable degree) this disadvantage.