基于设备固有噪声的置换语音信号检测

随着音频编辑软件的广泛使用,数字语音材料很容易被重新编辑。如何对数字语音信号原始性进行有效的认证成为一项急需解决的课题。提出了一种置换语音信号的小波包-谱相减（WPT-SS）的被动检测方法,利用语音设备固有噪声特性,对置换语音信号进行有效检测和置换区域定位,通过实验证实了该方法的有效性。     

多信道合作感知中基于信道可用概率的感知信道集合选择算法

首先分析了在给定感知信道集合和相应的可用概率集合条件下认知无线网络最大吞吐量的求解算法,接着给出了授权信道可用概率的估计方法,并在此基础上提出了一种基于授权信道可用概率估计的感知信道集合的次优选择算法。从分析结果与仿真结果可知,该次优选择算法与最优选择算法的性能差别不大,但是复杂度却大大降低了,另外该算法与已有算法相比可以得到更高的系统吞吐量。     

大北欧：携蓝牙耳麦向统一通信领域进军

统一通信（Unified Communication,简称UC）系统将语音、传真、电子邮件、移动短消息和多媒体数据等丰富信息集合为一体,可用固定电话、传真、手机、PC、掌上电脑等通信设备中的任何一种接收,在有线、无线、互联网之间架构起一个信息互联通道;而多功能耳麦在传统的固定电话、移动电话和个人电脑之间架起了一座桥梁,保证用户在UC平台实现无缝切换。     

语谱图傅里叶变换的二字汉语词汇语音识别

以语音信号的语谱图作为处理对象,提出一种基于宽窄带语谱图傅里叶变换频域图像二进宽度分带投影特征融合的二字汉语词汇语音识别算法。首先,对宽窄语谱图傅里叶变换频域图的图像意义以及相应的语音特性进行分析;然后,分别对宽窄带语谱图频域图像进行二进宽度分带列投影和行投影,将投影值作为语音识别的第一个特征参数集合和第二个特征参数集合,将以上两个特征集进行特征融合作为二字词汇语音识别的特征量,以支持向量机为分类器实现二字汉语词汇语音识别。实验结果表明,该方法对特定人二字汉语词汇语音的识别率可达96.8%,对非特定人二字汉语词汇语音的识别率可达98.8%,为解决汉语词汇整体语音识别提供了一种新的思路。     

PMCAP/CDMA协议在语音与数据混合业务中的应用研究

该文将CDMA移动通信系统上行信道中的一种MAC协议PMCAP／CDMA协议应用到语音与数据混合业务的情况，对协议建立数学模型，并对性能进行理论计算和仿真。为了保证对语音请求的优先分配，将PN码集分为语音可用码集，数据可用码集及语音与数据的预防码集。提出了新的动态码集分配方案。仿真表现，该方案较固定码集方案更好地提高了系统的综合性能，而负指数码字分配方案提供了语音与数据性能的很好折衷。     

新一代网络平台──CG6000C

3．可用的通道类型Fusion提供了一个事先定义的通道类型集合。这些通道的创建独立于MSP端点,用于创建两个端点之间的连接。一旦通道建立,就可以通过向通道发出命令来改变其工作模式。可用的预定义通道类型包括：1）G．7112）G．7231／AwithAnnexA(voiceactivitydetection）3）G729A／BwithAnnexB（Voiceactivitydetection）4）G．7265）T38faa图1示意一个典型的Fusion媒体流连接,通过一个全双工的G.711语音通道创建,连接一个DSO端点（PSTN接口）和一个RTP端点（IP网络接口）。因为该通道是全双工的,所以它执行编码或解码…     

置换的性质及选择

随机置换或置乱排列,在密码学中扮演着极其重要的角色。但是,并非所有的置换都有满意的置乱效果,能为我们所用。那么,哪些是可用的呢?又怎样来选择呢?本文在研究置换不动点,自然序码型以及漂移值特性的基础上给出了一般的取舍原则,提出了置换“合用”和“堪用”两级标准,并讨论了逆置换和可用性的关系。     

IP融合通信， 电力企业新选择

随着广大电力企业IP数据网络的完善及IP集合通信技术的成熟，在统一的IP网络上实现语音、视频、数据已经成为不可逆转的趋势。随着下一代企业网（NGeN）的兴起，语音、视频和数据三种通信手段进一步融合，以IP集合通信的形态开始服务于广大电力用户。本文试图分析电力网络目前的现状，对未来电力企业网络发展趋势进行深层次探讨。     

340秒放音电路

APULS近日推出了340秒OTP(一次性编程录入)放音电路,它可分成256段语音。可用手工控制,也可用微处理器控制。可以用在语音讲解、语音授课、语音导游、幼儿教育、多点语音报警系统等领域。本系列共有341/170/085三种型号,分别对应三挡放音时间。性能简介1、341/170/085分别采用8M/4M/2MEPROM存储器,上述允许放音的时间是在6kHz采样频率下。若要求高音质,需提高采样率,但时间容量会缩短。一般用于语音的采样率采用11kHz,允许时间将会减少45%。2、它采用4位ADPCM压缩算法合成语音,也可选用PCM算法(高音质,但放音时间短)。3、它有多…     

基于伪随机置换的朋友发现系统

近年来,随着移动社交软件的迅速发展,便携式移动终端已经渗透到人们生活的方方面面。在这些软件中,人们经常用到其中的一项功能—朋友发现,但是目前这个功能对于用户并不安全,大量隐私信息被云服务商所获取。针对此现象,文章基于目前隐私集合比较的现状,运用伪随机置换和安全多方计算协议并加以改进,设计出基于伪随机置换的朋友发现系统,本系统使用户找到他们集合的交集而且不泄露除交集以外的信息,具有一定推广价值。     

Noisy voice detection algorithm based on feature stream fusion

Aiming at the problem that the initial detection performance of voice segment was poor,and the voice continuity structure was damaged in voice communication,a noisy voice detection algorithm based on feature stream fusion was proposed.Firstly,the time domain feature stream,the spectral pattern feature stream and the statistical feature stream were extracted according to the voice characteristics.Secondly,the voice segment in the noisy audio was estimated by different voice feature streams.Finally,the voice prediction probability obtained by each feature stream was weighted and fused,and the voice estimation probability was processed in short time by the hidden Markov model.The performance test of composite voice database under the condition of multi-type noise and different signal-to-noise ratio shows that compared with the baseline model based on Bayesian and DNN classifier,the voice detection accuracy of the proposed algorithm is improved by 21.26% and 11.01% respectively,and the quality of target voice is significantly improved.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

多用户远程语音交互终端接入设备设计与实现

基于低空飞行航空管制组网话音通信应用背景,从音频接口、模数/数模转换接口和网络接口3个核心处理模块出发,研究了网络化语音接入设备的硬件关键技术。从C语言级和汇编级两方面对G.729A语音编解码算法进行改进优化,并给出了语音数据的处理流程,以单片TMS320C6455DSP实时实现了语音4路压缩编码和20路解码处理。采用TCP/IP网络协议实现了多路语音网络通信功能,分析了多路语音串扰问题并设计了相应的解决方案。测试表明,基于该技术实现的设备语音处理延迟小于10 ms,满足多用户语音实时传输要求。     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Quantitative investigation of the vibration pattern of the substitute voice generator

After a total excision of the larynx, mucosal tissue at the upper part of the esophagus can be used as a substitute voice generating element. The properties of the tissue dynamics are closely related to the substitute voice quality. The process of substitute voice is investigated by recording simultaneously the acoustic signal with a microphone and the vibrations of the voice generator with a digital high-speed camera. We propose an automatic image-processing technique which is applied to analyze the vibration pattern of the substitute voice generating element. First, an initialization step detects the voice generator within a high-speed sequence. Second, a combination of a threshold technique and an active contour algorithm tracks the tissue deformations of the substitute voice generator. The applicability of the algorithm is shown in three high-speed recordings. For the first time, tissue deformations of substitute voice generating elements are successfully tracked. The results of the image processing procedure are used to describe quantitatively the temporal properties of the substitute voice generator. Comparisons of the spectral components of tissue deformations and tracheoesophageal voice signals reveal the close relationship between the vibration pattern of the substitute voice generator and the quality of substitute voice.     

语音伪造及检测技术研究综述

语音承载着人类语言和说话人身份信息,通过语音伪造技术可以精确模仿目标说话人的声音以达到欺骗人或机器听觉的目的。目前,深度伪造（Deepfake）正在对全球的政治经济及社会稳定带来极大的威胁,其中语音伪造是Deepfake实现舆论操控的核心技术之一。近年来语音伪造技术在拟人度、自然度方面有了显著进步,使得语音伪造检测技术面临着更大的挑战。本文对当前主流的语音伪造和伪造语音检测技术研究现状进行综述,主要包括:1）对主流语音伪造技术,包括语音合成、语音转换和语音对抗样本的基本概念、技术发展历程和研究进展进行综述;2）对伪造语音检测技术的基本概念、性能评价指标、主要技术实现原理和性能效果进行综述;3）对伪造语音检测相关的主流竞赛、常用数据集和可用代码工具资源进行介绍;最后对语音伪造和检测技术现存的挑战性问题和未来的研究方向进行讨论。      

AAL2分组话音系统信元装配时延的研究

该文研究将AAL2分组填入ATM信元载荷域时的信元装配时延。得出结论:ATM信元装配时延由话音源编码速率、分组占用时长以及接入AAL2分组话音复接器中的话音源个数确定。当话音源编码速率较低,接入AAL2分组话音复接器中的话音源数较小时,信元装配时延可能很大,需要设置定时器以限制信元装配时延,例如当话音源编码速率为8kb/s时,可令定时器的取值为3ms;当话音源编码速率为32kb/s时,若分组占用时长为5ms,一般无需使用定时器。     

短波地空语音组网中VoIP语音质量评估

短波地空话音组网是短波话音通信发展的必然趋势,组网后VoIP技术的引入会为短波语音通信带来很大的便利,但同时语音质量会受到影响,其中VoIP编码技术是影响语音质量的主要因素之一。通过Matlab对采用不同编码技术重构后的语音,经过基于Watterson模型的短波信道传输后到达接收端的质量进行了PESQ评估,并仿真分析了丢包率对不同编码语音质量的影响,得出了不同编码的优劣性。     

动态多通道卫星话音／数据综合的S－ALOHA性能分析

本文分析了一种新的多通道话音／数据综合ＶＳＡＴ系统的Ｓ－ＡＬＯＨＡ性能，并给出了数值计算结果；从中可看到该协议既能提供较高的吞吐量，又能有效地减小话音时延，并能协调话音和数据之间对性能要求的差异，文中考虑了Ｓ－ＡＬＯＨＡ中分组达到时间抖动的影响．文中还讨论了话音在重负载的条件下可占用数据通道的动态分配方案，并给出了动态分配的算法。     

Asynchronous voice: a personal account

Asynchronous voice is the interactive communication process of people leaving voice messages for other people and the other people responding with their voice messages. A primitive form of asynchronous voice is a kind of telephone tag in which people use voice mail to have an interactive conversation. The author gives a personal account of his work with asynchronous voice and asynchronous learning.