该系统中涉及到的主要技术是语音编解码技术。
同时通过自制一块语音编解码拓展板，实现了全双工通信。
Meanwhile, an audio codec circuit board is made to realize full-duplex communication.
研制的语音编解码系统板经测试，各项功能均符合设计要求。
After testing the voice codec system, all the functions have met the design requirements.
最后分析了此多通道实时语音编解码方案所需的存储空间和计算复杂度。
By exploiting the multichannel ADCs and PWMs of the TMS320F2812, the codec is extended to a multichannel real-time speech codec.
该优化设计方法对其它语音编解码标准的DSP实现具有较大的参考价值。
The optimization method proposed in this paper is valuable to the DSP implementation of other speech coding standards.
介绍了语音编解码算法原理、声码器系统的硬件结构、工作流程以及软件实现与代码优化。
The speech coding algorithm, the hardware architecture and work flow of the vocoder system, software implementation and code optimization are introduced.
经过较长时间的监测运行，G . 726语音编解码系统能稳定工作，实时采集和回放语音。
After a long time monitored operation, G. 726 speech codec system can perform stable work, real-time data acquisition, and playback of voice.
介绍了自适应多码率语音编解码算法及其基于TMS320F 2812定点DSP芯片的实现方案。
An Implementation of the adaptive multi-rate speech codec based on the TMS320F2812 fixed-point DSP chip is presented.
本课题从现代通信实验系统的要求出发，设计并实现了基于DSP(数字信号处理器)的语音编解码系统。
Starting from the request of the modern communication experimental system, the subject have designed and implemented a voice codec system based on DSP (digital signal processor).
现代语音编解码技术中，除了单纯的编码解码之外，针对传输媒质、环境等问题，又涌出了一系列与之相关的技术。
Except simple coding in modern speech coding technology, there is a serial of technology to solve transmitting medium and surrounding problems.
测试结果表明所设计的语音编解码器能够实时完成四路语音记录，具有较高的恢复语音质量，性能指标达到设计要求。
Experiments proved that the codec can recode four channels speech in real time and the quality of the recovered speech is satisfying. And the design intention is realized.
最后讨论了基于上述硬件系统的软件设计，主要分为三个部分：dsp的初始化程序、中断服务程序和语音编解码程序。
Finally, this paper discussed the software system design based on above hardware platform, including the DSP initialization program, the interrupt request service and speech coding.
它不是单一的一种技术，而是综合了通讯网络的多种技术在内的一个系统，包括了信令协议、语音编解码、媒体传输和网关控制。
It is not a single but a system integrated with several technologies constructing communication network, including Signaling Protocol, Voice Codec, Media Transport and Gateway Control.
在众多语音压缩标准中，由itu - T(国际电信联盟)于1996年制定的G. 729语音编解码标准表现得十分抢眼。
Among so many voice compression standards, the performance of g. 729 voice codec which is established standards by ITU-T (International Telecommunication Union) in 1996 is very eye-catching.
文章参照3gppTS协议，对于自适应多速率(amr)语音编解码算法进行了深入的研究，并且对其工作原理做了详细的分析。
This paper, referring to 3gpp TS, gives an in-depth study on AMR, and carries out a detailed analysis on the working principle of AMR.
在对实现的TETRA语音编解码算法进行性能测试、分析的基础上，文章提出了在TETRA语音编码的预处理部分采用优化算法。
On the basis of the performance testing of our real-time codec is optimum algorithm introduced into pre-processing of the speech encoder of TETRA.
在各种语音编解码改进算法的实现过程中，我们采用了TI公司的TMS320C 5402 DS K板为硬件平台及CCS集成开发环境。
In the realization of these kinds of mended arithmetics, the TMS320C5402 DSK of ti company is adopted as the hardware flat roof and the CCS integrated exploiture environment.
设计中采用了DSP、低比特率语音压缩编解码、信道复用、快闪存储、串口和调制解调器通信等技术。
In this design many technologies such as DSP, low bit-rate speech compression, channel multiplexing, Flash Memory and communication with modem through se rial port are used.
利用媒体网关进行语音流的编解码，确保PSTN网和IP网络间的互通性；
Media gateway is used to encode and decode voice steam and connect PSTN network to the IP network.
研究了陆地集群无线通信（TETRA）数字集群移动通信系统的语音压缩编解码算法的基本原理。
This paper describes the basic theory of speech compression encoding and decoding algorithm in TETRA (terrestrial trunked radio) system.
语音编码的基本问题就是在给定编码速率的条件下如何得到尽可能好的重建语音质量并保证尽可能小的编解码时延和适当的运算复杂度。
The fundamental of speech coding is the fact that how to get better quality of reconstructed speech and ensure the less time delay and proper operational complexity at the given coding rate.
本发明所述的装置是在主控板中增加一个语音模块。语音模块具有信号监测功能，编解码功能，拨号信号发送功能。
The apparatus adds a voice module in the main control board, which has signal monitoring function, encoding and decoding function, dialing signal sending function.
本文将语音增强的算法和低信噪比下的端点检测算法应用到CELP编解码器中，提出了EV - CELP编码模型。
In this paper, the speech enhancement algorithm and the new endpoint detection algorithm are applied to CELP encoder, and EV-CELP model is presented.
本文将语音增强的算法和低信噪比下的端点检测算法应用到CELP编解码器中，提出了EV - CELP编码模型。
In this paper, the speech enhancement algorithm and the new endpoint detection algorithm are applied to CELP encoder, and EV-CELP model is presented.
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