TETRA与G.729语音编码算法的比较研究

为了研究TETRA中的ACELP与ITU-TG.729中CS-ACELP两种语音压缩编码算法的异同,分别就以下几个方面对这两种算法进行了比较:预处理、线性预测分析(LPC)、自适应码本的搜索和固定码本的搜索.然后在VC++6.0环境下实现了这两种算法,并对算法的仿真结果进行了对比分析.最后采用ITU-TP.862.1标准感知语音质量评价(PESQ_LQO)对两种算法合成语音的质量进行客观测评,测试结果表明,G.729合成语音的质量比TETRA合成语音的质量要好.     

A perceptual quality model intended for adaptive VoIP applications

Quality models predict the perceptual quality of services as they calculate subjective ratings from measured parameters. In this article, we present a new quality model that evaluates Voice over IP (VoIP) telephone calls. In addition to packet loss rate, coding mode and delay, it takes into account the impairments due to changes in the transmission configuration (e.g. switching the coding mode or re‐scheduling the playout time). Moreover, this model can be used at run time to control the transmission of such calls. It is also computationally efficient and open source. To demonstrate the potential of our model, we apply it to select the ideal coding and packet rate in bandwidth‐limited environments. Furthermore, we decide, based on model predictions, whether to delay the playout of speech frames after delay spikes. Delay spikes often occur after congestion and cause packets to arrive too late. We show a considerable improvement in perceptual speech quality if our model is applied to control VoIP transmissions. Copyright © 2005 John Wiley & Sons, Ltd.     

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.     

一种基于感知语音质量评价的列车显示器语音自动检测方法

列车显示器语音测试时,调试员需要通过耳朵判断语音的质量,而调试员对语音的判断会受环境噪声、显示器装配等因素影响,造成了出厂的列车显示器语音质量存在不一致性的现象.对此,文章提出一种语音自动检测方法,其利用麦克风采集列车显示器发出的声音,采用感知语音质量评价(PESQ)算法对列车显示器发出的语音进行自动检测和评价.仿真结...     

基于E1接口感知语音质量测试设备的实现方案

语音音质评价从主体上可分为主观评价和客观评价两种。由于音质好坏最终取决于人的主观感受,所以在语音系统中主要采用主观评价的方法。但是这种方法费时费力,同时受到测试条件和测试人员主观因素的影响,降低了测试结果的可靠性。针对上述缺点,设计了一种客观评价语音音质的测试设备实现方案,该测试设备基于E1接口,采用ITU-T P.862 PESQ算法模型。     

突发大时延下的IP语音缓冲算法

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的情况,从而难以获得好的语音质量。对于这个问题,提出针对突发大时延存在下的自适应语音缓冲算法。算法通过控制语音包在语音缓冲队列中的位置来控制语音包的播放时间,从而可以尽量减小语音裂缝(Gap)的出现。算法将突发大时延存在下的最大丢包率可以扩大到20％,而一般的预测算法只能容忍5-10％的最大丢包率。通过基于听觉模型的客观音质评价(PESQ)仿真计算,以及实际应用表明本文的算法对有突发大时延存在的网络中的语音通信质量有一定的改善作用。     

P2P VoIP应用的性能评测

基于IP技术的语音分组传输(VoIP)电话目前被广泛使用,Skype与GTalk是VoIP应用的两个典型代表.在可控网络环境下,通过调整信道容量、时延、丢包、抖动等网络参数,利用PESQ MOS方法评测了Skype与GTalk的语音质量,并且讨论了在可变网络环境下的动态适应性策略.     

具有后向基音检测的8 Kbit/s LD-aCELP语音编码

以降低码率为目的对G.728算法进行改进,提出了一个延迟为2.5 ms的8 Kbit/s的语音编码算法。算法引入了由最近的历史激励构成的自适应码书和归一化的固定码书的双码书结构。计算增益真值并量化,增益量化时对自适应码书用固定量化,固定码书用自适应量化。码书搜索时先进行后向基音检测,在基音周期T附近对自适应码书进行精细搜索。搜索64个自适应码矢、256个固定码矢和各自8个增益值获得最佳激励,每帧耗费20 bit。用平均分段信噪比和感知语音质量评价（PESQ）测试,改进算法编码质量接近于G.728。     

A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems

Jitter buffer plays an important role in Voice over IP (VoIP) applications because it provides a key mechanism for achieving good speech quality to meet technical and commercial requirements. The main objective of this paper is to propose a new, simple-to-use jitter buffer algorithm as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance, in terms of enhanced user-perceived speech quality and reduced end-to-end delay. Supported by signal processing features, the new algorithm, the so-called Play Late Algorithm, alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. The results show that the new algorithm achieves the best performance under different network conditions when compared to conventional static and adaptive jitter buffer algorithms. The results reported here are based on live tests and emulated network conditions on real mobile phone prototypes. The mobile phone prototypes use AMR codec and support full IP/UDP/RTP stack with IPSec function in some of the tests. The method for perceived speech quality measurement is based on the ITU-T standard for speech quality evaluation (PESQ).

西门子GSM设备增强型全速率信道研究

增强型全速率信道(EFR)通过改进信道编解码算法能有效地提高GSM网络用户的实际语音通话质量.主要对西门子GSM设备的EFR开通方法以及开通后的对比测试进行了研究和分析,为提高GSM网络语音质量提供了新思路.