单片机实现的数码录放机

采用单片机作为主控核心，利用TP3057芯片实现语音的编／解码，D6571芯处实现模／数转换和压缩以及数／模转换，采用Flash ROM存储技术，实现了语音的存储与回放功能。介绍了语音存储与回放系统的原理，硬件结构和程序设计。     

基于以太网的串行数字语音通信系统

针对当前广播对讲系统通信带宽受限、音质效果一般和扩展性差等问题,在对现有广播对讲系统进行分析比较的基础上,设计一种基于以太网的串行数字语音通信系统。阐述了系统的结构及软硬件设计,采用内嵌ARM内核的以太网控制器W7500P作为主控芯片,W5500和主控芯片构成双网络接口结构,同时使用语音编解码芯片VS1063实现模拟语音信号与数字音频信号的转换以及MP3格式数字音频信号的编解码功能,从而在保证语音质量的前提下降低语音信号对通信带宽的要求,以实现语音信号的高效传输。该系统连接简单、布线方便、成本低廉。实验测试表明,系统的语音质量、实时性和可靠性均符合现场使用的要求。     

多速率语音芯片AMBE-2020的开发和应用

AMBE-2020是由DVSI公司推出的高性能多速率语音编码、解码芯片，它的语音算法采用专利的AMBE语音压缩技术。该芯片能在较低的速率下实现全双工、高质量的语音通信，并且操作简单、编码速率可变、功耗小等优点。介绍了AMBE-2020芯片，并建立了一个简单的语音压缩系统的应用实例。     

基于USB2.0的语音数据采集系统设计

本文提出并设计实现了基于USB2．0的语音数据采集系统，该系统以TMS320VC5402 DSP芯片为主控机，采用USB2．0协议芯片ISP1581实现系统与计算机之间的高速串行数据传输，重点介绍了USB设备主从两端的软硬件设计方案。     

基于AMBE-3000声码器的语音通信系统设计

介绍了语音系统组成和AMBE-3000声玛器的基本原理,对基于AMBE-3000声码器语音系统的实现过程进行了详细阐述。本系统以FPGA为核心,利用全双工AMBE-3000声码器,实现了语音数据的双向数据传输。一方面FPGA作为主控芯片,分别对TLV320AIC32和AMBE-3000芯片的功能进行配置和控制;另一方面,FPGA实现TLV320AIC和AMBE-3000芯片之间的数据交互,并对数据进行处理。该系统在8 kHz低速、纯话音应用环境下,能够获得更好的语音质量,而且性能优于G.729。     

潜水员水下近距离简易通信系统的设计

介绍了基于单片机的潜水员水下简易通信系统的设计与实现,给出了它的硬件原理图和软件程序框图。该系统采用短消息通信,并且通过语音播放收到的消息。硬件部分采用ATMEL公司的AT89S52作为主控芯片,实现了FSK调制解调,并控制ISD公司的ISD4002语音芯片进行话音的录放。系统利用水下电流场,完成了潜水员间的水下近距离简易通信。     

基于MC68360视频压缩系统的设计分析

提出了一种新的嵌入式视音频压缩/解压缩系统的设计方案,该方案采用MC68360作为主控芯片,VW2010作为解压缩核心芯片.实验结果证明该方案具有可靠性高、抗干扰能力强、功耗小、性价比高等优点,可以很好地实现MPEG4标准的视音频压缩/解压缩.     

基于CAN总线的煤矿语音通信系统的设计

针对煤矿语音通信系统的现状,设计了一种能够满足煤矿远程语音通信的煤矿语音通信系统。该系统采用自带CAN控制器的PIC18F458作为主控芯片,AMBE2000作为语言编解码芯片,以CAN通信作为通信模式。详细介绍了系统的软硬件设计。通过实验证明该系统的声强、失真度、可懂度等指标均符合现场的需求。     

基于GSM的红外遥控语音提醒器的研制

在此设计了一种基于GSM的红外遥控语音提醒系统,该系统采用STC89C52单片机为主控芯片,并使用红外遥控控制整个系统。系统中ISD4004语音模块可在用户设定的时间播放预先录制的语音信息。实现了将GSM技术融入到设计中,用户即使不在语音提醒器的附近,一样可以通过手机接收到语音提醒。创新点是打破以往提醒系统的提醒范围的局限性,极大地提高了提醒系统的实用性和可靠性。     

基于TMS320C54x DSKplus的实时TCM语音编解码算法的实现

描述了基于TMS320C54x数字信号处理器的TCM语音压缩编码系统。该系统是在TMS320C54xDSP入门套件（DSK,DSP Starter Kit）板上实现,充分发挥了芯片的专用硬件逻辑、专业化的指令以及板上TLC320AC01模拟接口语音处理系统。有效而快速地完成了TCM语音压缩系统的模拟,并给出相应的实验结果。     

Personalized voice activated grasping system for a robotic exoskeleton glove

This paper proposes a novel human machine interface (HMI) and electronics system design to control a rehabilitation robotic exoskeleton glove. Such system can be activated with the user’s voice, take voice commands as input, recognize the command and perform biometric authentication in real-time with limited computing power, and execute the command on the exoskeleton. The electronics design is a stand-alone plug-and-play modulated design independent of the exoskeleton design. This personalized voice activated grasping system achieves better wearability, lower latency, and improved security than any existing exoskeleton glove control system.     

数字对讲机中语音编码的研究与DSP实现

根据TDD数字对讲机的特点,以TMS320VC5509A为平台,结合微处理芯片MSP430和射频芯片CC1100,提出了一种数字对讲机系统的结构设计方案,该设备主要应用于抢险、救灾、野外作业等缺乏基础通信设施的环境。设计方案中,利用TMS320VC5509A实现无线语音通信中广泛采用的G.729A语音压缩编码,获得了较低的编码速率和较好的话音质量,在满足话音通信实时性要求的同时,提高了频谱的利用率。     

Efficiency of packet voice with deterministic delay

Packet switching is appealing for carrying real-time traffic because it can benefit from (possibly variable bit rate) compression schemes and statistical multiplexing to more efficiently exploit network resources. This work explores the efficiency of IP telephony in terms of the volume of voice traffic carried with deterministically guaranteed quality related to the amount of network resources used. An IP network carrying compressed voice is compared to circuit switching carrying PCM (64 kb/s) encoded voice, and some design choices affecting IP telephony efficiency are discussed     

基于以太网的话音和数据系统的实现

介绍了一种基于以太网的话音和数据通信系统的实现。该系统由一个高级终端、若干普通终端和一个以太网交换机组成。高级终端和普通终端都支持话音、非IP数据和IP数据。各种业务之间相互独立,互不干扰。终端的硬件结构基于MPC860T、DSP和FPGA,软件采用模块化设计,使用VxWorks实时多任务操作系统,采用广播技术和话音叠加实现多个终端间的电话会议功能。     

基于FPGA的语音录制与回放系统

系统用FPGA实现了I2C总线控制器,以Altera公司的NiosⅡ嵌入式软处理器为核心,结合高品质数字信号音频编/解码芯片WM8731成功地实现了语音的录制及回放功能,同时利用Matlab7.0.4软件对所采集的语音数据进行仿真。系统采用SoPC技术,自行设计采集模块和I2C协议驱动模块,并通过AWALON总线挂载在Nios软核上实时高速采集与回放。实践表明,系统具有集成度高,稳定性好,实时性强的特点。     

Latency-driven distribution: infrastructure needs of participatory entertainment applications

This article evaluates network and server infrastructure requirements to support real-time flows associated with networked entertainment applications. These include the state information flow to update the status of the virtual environment and immersive communication flows such as voice, video, gesture, and haptics communication. The article demonstrates that scaling these applications to large geographical spreads of participants requires distribution of computation to meet the latency constraints of the applications. This latency-driven distribution of computation is essential even when there are no limitations on the availability of computational resources in one location. The article provides detailed results on distributed server architectures for two of these real-time flows, state information and immersive voice communication. It also identifies a generic set of requirements for the underlying network and server infrastructure to support these applications and propose a new design, called switched overlay networks, for this purpose.     

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。     

Real-time voice over packet-switched networks

We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics     

基于ARM的家用智能垃圾桶设计

以ARM11 S3C6410为核心设计了一种家用智能垃圾桶,用拾音器作为声音传感器,采用延时估计法实现声源方位的实时检测,从而实现语音控制垃圾桶的运动;采用红外传感器实现垃圾桶行进的蔽障功能;同时,采用语音识别技术实现用户对垃圾桶的前、后、左、右行驶或开启、关闭垃圾桶盖等各种语音指示的识别。从而实现垃圾桶的智能化与人性化,给生活带来便利。     

Admission control scheme for soft handoff in DS-CDMA cellular systems supporting voice and stream-type data services

The soft handoff call requests of real-time services in third-generation (3G) direct-sequence code-division multiple access (DS-CDMA) and first- and second-generation cellular systems are more important than new call requests from the viewpoint of quality of service (QoS). Rejection of soft handoff requests causes forced termination of an ongoing real-time call, which is a severer problem than blocking of new call attempts. An admission control scheme that can guarantee a higher QoS for the soft handoff requests of real-time services in 3G DS-CDMA systems is proposed for delay-sensitive voice and delay-tolerant stream-type data services. The proposed scheme (P-Scheme) accommodates both voice and data services by utilizing the full bandwidth. However, voice soft handoff call requests are given priority over new voice call and stream-type data packet requests by suppressing interference from stream-type data services according to voice soft handoff requests, and by varying interference levels. Performance of the P-Scheme is evaluated using a Markovian model. Results are compared with a conventional reservation scheme (C-Scheme) that reserves resources exclusively for voice soft handoff requests. Numerical results show that system performance can be significantly improved using the proposed P-Scheme, compared with the conventional C-Scheme, when various types of service are supported in third-generation DS-CDMA systems.