用AC4830xC-C和TCM38C17实现四路语音编解码系统

AC4830xC-C是美国.AudioCodes公司生产的语音专用芯片，它支持多种码率的语音编解码国际标准，同时可提供传真和数据中继功能。而TCM38C17则是美国TI公司的语音PCM编码芯片。文章介绍了两种芯片的基本性能及工作原理，并给出了一种基于这两种芯片设计的四路语音编解码系统的实现方案。     

语音编码与语音芯片

本文概述了语音编码，着重介绍了语音录放芯片Ｔ６６６８和单片可变速率声码Ｑ４４００，其中给出了一个车载数字语音喊话器的实例。     

基于AMBE-2020的可编程语音压缩编码系统

介绍了语音双工编解码芯片AMBE-2020的功能特点,给出了一种基于AMBE-2020和FPGA(现场可编程门阵列)的语音压缩编码系统的具体设计与实现,阐述了在设计过程中一些需要注意的问题.该系统具有设计简单、语音编码速率可变、音质好、功耗低等优点,可广泛应用于各种语音通信.     

语音编码技术进展

语音编码技术进展樊昌信一、语音编码的目的和应用语音编码是将模拟语音信号数字化的手段。语音信号数字化后，可以作为数字数据来传输、存储或处理，因而具有一般数字信号的优点。在实际应用中，这些优点是很重要的。１．数字语音信号在经过信道传输时，信道引入的噪声和...     

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。     

AMBE－1000语音编码－解码器芯片及其应用

AMBE－1000是由Digital Voice System.Inc推出的高性能多速率语音编码－解码芯片,它的语音编码－解码算法采用专利语音编码技术AMBE,具有语音音质好的编码波特率低、功耗低优点。可广泛应用在数字语音通讯、调废语音记录等场合。本文介绍了AMBE－1000的结构、引脚定主、数据格式和设计要点,并给出了AMBE－1000用于机车调度录音装置的应用实例。     

低速率语音编码技术

本文以ITU最新公布的语音编码方案MP MLQ /ACELP为例 ,介绍低速率语音编码技术的发展 ,并将该方案和GSM系统的全速率语音编码方案进行比较 ,说明中低速率语音编码在技术和应用上的一些特点。     

多媒体通信中的语音编码及其DSP实现

介绍了各种语音编码方法、讨论了多媒体通信中的语音编码，详细介绍了G．729的编解码方法及如何用DSP芯片来实现语音编码。     

多速率语音芯片AMBE-2000的开发和应用

AMBE-2000是由Digital Voice System．Inc推出的高性能多速率语音编码芯片，具有语音音质好、码波特率低、使用灵活性大等优点。介绍了AMBE-2000的使用，在构建的通信平台上进行了大量的软硬件实验和测试。最后总结出该芯片开发的关键技术并用以指导实践。     

专用可编程语音编码电路的设计与实现

在人工电子耳蜗系统中，专用可编程语音编码电路实现将语音特征信息包容错编码为传输脉冲，并按要求的传输格式发送编码脉冲。电路独立的指令系统保证了电路能够灵活实现系统所需的各种数据传送格式。电路的容错编码能力保证了传输数据有较强的抗干扰能力。本文介绍了该电路的功能、电路的指令集设计、电路结构和逻辑设计方法以及电路的版图设计过程，并讨论了电路芯片的测试结果。电路版图面积为４．５ｍｍ×４．５ｍｍ，包含１０００个等效门及４３个引脚。     

IP电话的语音压缩技术

本首先介绍了语音编码技术中对语音质量的评价方法，然后对几种典型的IP电话语音压缩技术标准进行了阐述，最后，对三种常用的IP电话语音压缩技术参数进行了比较。     

Voice packetization and compression in broadband ATM networks

Some methods of supporting voice in broadband ISDN, (B-ISDN) asynchronous transfer mode (ATM), including voice compression, are examined. Techniques for voice compression with variable-length packet format at DS1 transmission rate, e.g., wideband packet technology (WPT), have been successfully implemented utilizing embedded adaptive differential pulse code modulation (ADPCM) coding, digital speech interpolation (DSI), and block-dropping schemes. For supporting voice in B-ISDN, voice compression techniques are considered that are similar to those used in WPT but with different packetization and congestion control methods designed for the fixed-length ATM protocol at high speeds. Possible approaches for packetization and implementation of variable-bit-rate voice coding schemes are described. ADPCM and DSI for voice coding and compression and cell discarding (CD) for congestion control are considered. The advantages of voice compression and CD in broadband ATM networks are demonstrated in terms of transmission bandwidth savings and resiliency of the network during congestion     

SSM2165语音压控电路的设计应用

短波通信发射机中频数字化之前,需要一个稳定的采样信号,我们通常需要设计一个语音压控电路。本文简述了一种语音压控电路在短波通信发射机中的基本作用,并介绍了语音压控芯片SSM2165的基本特性。最后结合实际应用,介绍一种基于该芯片的语音压控电路应用及其使用效果。     

基于DSP的语音信号的采集压缩与数据传输

文章基于DSP的语音系统,采用TMS320VC5402作为微控制器,实现了对语音信号的采集压缩和传输数据等功能.通过自设压缩算法库将采样信号进行压缩存储,通过串行通信接口SPI完成数据发送和接收.     

数字语音的电力线载波全双工通信系统设计

语音信号的电力线载波传输已经有了广泛的应用,但均是以模拟方式来传输信号的,故通信质量不高,抗干扰能力差,以数字信号作为语音传输方式,就可以克服上述缺点.以新型语音压缩编解码器AMBE2000和电力载波扩频调制解调芯片SC11128为基础的设计,就可以使当前先进的语音压缩技术以及基于恶劣电力网环境的低压电力线载波技术得到充分的应用,结合单片机对电路状态的控制以及双口RAM对数据的处理从而实现了电力线信道的全双工数字语音通信.     

语音数字信号压缩技术在TES系统中的应用

介绍ADPCM标准、RLPC编码原理,编、解码器方框图及工作过程。民航卫星通信网TES系统为节省卫星转发器频率资源对传输的语音信号进行压缩处理,其信道单元基带信号处理器对语音信号进行CCITT推荐的G.721-ADPCM编码和修斯公司专利技术开发的RLPC编码处理,将64 kb/s语音数字信号压缩至32 kb/s,16 kb/s,9.6 kb/s传输,实现语音质量满足一般通信要求的低速率语音信号传输。     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

Voice quality evaluation in wireless packet communication systems: a tutorial and performance results for RHC

As wireless systems evolve toward supporting a wide array of services, including traditional voice service, using packet-switched transport, it becomes increasingly important to assess the impact of packet-switched transport protocols on voice quality, in this article we present a tutorial on voice quality evaluation for wireless packet-switched systems. We introduce an evaluation methodology that combines elementary objective voice quality metrics with a frame synchronization mechanism. The methodology allows networking researchers to conduct effective and accurate quality evaluation of packet voice. To illustrate the use of the described evaluation methodology and interpretation of the results, we conduct a case study of the impact of robust header compression (ROHC) on the voice quality achieved with real-time transmission of GSM encoded voice over a wireless link.     

Header compressed VoIP in IEEE 802.11

Header compression techniques such as robust header compression can be used to reduce the overhead of IP-based traffic. Voice over IP may replace voice circuits in the next generations of wireless networks, and it is the type of traffic that benefits most from header compression because its packets have small payloads. IEEE 802.11 is a technology that will play an important role in the next generations of wireless networks. The study reported in this article shows that the maximum gain of the RoHC?s U-mode when applied to VoIP over IEEE 802.11 is about 23 percent for medium or better voice quality. Values for the RoHC Umode parameters over IEEE 802.11 are also suggested.