最后对该声码器进行性能分析。
一个默认地噪声码列表可供每个主要的欧洲语言使用。
A default list of noise words is supplied for each major European language.
您可以更改这个噪声码的供应列表或者定义您自己的噪声码。
You can alter the supplied list of noise words or define your own.
要获得更多关于噪声码的信息，可以参考oracle文档。
For more information on noise words, check the Oracle documentation.
在低速率声码器中，对激励信号的描述直接影响重建语音的质量。
The description precision of the excitation signal greatly influences the quality of reconstructed speech in low bitrate vocoders.
干扰字：当您创建索引时，您需要考虑Oracle所称的“噪声码。”
Noise words: When you create indexes, you need to consider what Oracle calls "noise words."
本文试图将语音增强技术与MBE模型相结合以提高声码器抗噪声的性能。
This thesis intends to improve the performance of IMBE speech vocoder under noisy enviroment by the combination of speech enhancement techniques with the MBE speech model.
特别说明了该声码器根据外部提供的同步时钟自适应地改变工作速率的方法。
Especialy realized the method of adapting to change its work rate in accordance with outward provided synchronization clock.
当您创建索引时，它们将会自动地利用一个默认地停止列表来忽略掉这些噪声码。
When you create indexes, they will automatically use a default stop list to ignore noise words.
论文设计了一个基于多带激励语音压缩编码算法的DSP系统-多速率声码器。
This paper describes a DSP system which based on multi-band excitation voice compression coding algorithm - multi-rate vocoder.
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.
介绍了应用ambe- 2000TM声码器芯片设计的语音通信系统的具体实现方案。
This paper introduces the design of a voice communication system based on the AMBE-2000TM vocoder chip.
主要介绍应用声码器基于网络系统设计的语音加密通信系统方案，给出了硬件系统组成结构。
This paper briefly described the voice encryption communication system design project which use sounder based on computer network. The hardware system constitution is shown in the paper.
LPC声码器在准确提取声道参数的情况下，激励模型的优劣直接影响着综合语音的质量。
When LPC vocoder exactly extracts vocal tract parameters, whether an excitation model is good of not will severoly affect the quality of the synthetic speech.
最后，着重讨论了在这种体系结构和控制流程下，系统对电路资源、声码器资源和音源的管理。
At last, it mainly discusses the management of circuit, vocoder and sound based on this architecture and calling control flow.
介绍了采用多cpu系统和VL SI声码器为核心的语音数据复接器，分析了系统的工作原理和特点。
At last, this paper introduces a speech and data multiplexed based on Multi-CPU and VLSI vocoders. The operating principle and characteristics of the multiplexed are analyzed.
码激励线性预测（CELP）编码在中低速率上与传统的LPC声码器相比可以提供更加自然的语音质量。
Code-Excited Linear Predictive (CELP) coding can provide natural speech quality at medium and low bit rates in contrast with conventional LPC vocoder.
在语音编码的应用环境中，特别是在军事应用中，强噪声环境下声码器性能的改进是一个亟待解决的问题。
In the environment of speech coding application, especially in military applications, the vocoders quality in noisy environment needs to be improved.
对面的坡地宛若印度教的一个巨具，从各个角度斜向太阳。斜坡上是他滑动脚步时留下的无声码似的印痕。
The opposing slopes, inclined at all angles to the sun like an immense Hindu yantra, were marked with the muffled ciphers left by his sliding feet.
在相同的速率下，VMR-WB语音声码器的主观质量和性能较现有的窄带和宽带编码标准都有明显的提高。
The subjective quality and performance of VMR-WB is superior to that of existing narrowband and wideband coding standards operating at the same bit rates.
MELP声码器采用混合脉冲和噪声激励解决了经典LPC的嗡嗡声的问题； 引入了抖动浊音状态以克服音调噪声；
MELP vocoders utilize mixed pulse and noise as the excitation to elimate the buzzes in traditional LPC vocoders, and add a jitter voicing state to overcome the tonal noise.
本文首先研究了AMR声码器的算法，AMR声码器主要由多速率语音编码器、源控速率方案和错误消除机制三大部分组成。
First the paper studies the algorithms of AMR speech coder. The AMR speech coder consists of the multi-rate speech coder, a source controlled rate scheme, and an error concealment mechanism.
分段声码器是一种高效的低速语音编码方法，针对分段声码器实现的关键技术语音分段方法进行了研究，提出了一种分段新算法。
Segment vocoder is an efficient very low bit rate speech coding algorithm. Speech segment method, one of the key technology of segment vocoder, is studied and a new algorithm is presented.
声码器是一种数字化语音软件，将演讲语音转变成信息包，因此它可以以电子方式传送，并在接收端进行解码，以便它可以被人听到。
A vocoder is software that digitizes speech, transforms it into information packets so it can be electronically transmitted, and decodes it at the receiver so that it can be heard as speech.
基于一定的解码状态，声码器通过最小均方误差（MMSE）估计的方法估计最优参数，充分降低信道误码对重建语音质量的影响。
The minimum mean square error (MMSE) is computed for each decoding state to estimate optimal parameters and to reduce the influence of the bit error.
实验结果表明：在1%以上的随机误码率下，采用本文提出的纠错 算法后，信道误码引起的声码器基音参数误差能够降低10%以上。
Tests show, the modified restoration scheme can reduce the pitch error due to channel errors at least 10% on channels with bit-error-rates above 1%.
若存cmg系统中的声码器单板上运行这种改进的EVRC编解码算法，在支持相同数目的EVRC编解码器的情况下，能为系统节省一定的DSP数量。
If this codec is utilized on the vocoder board in the CMG system, with the same number of EVRC sustained, we can reduce quite a few DSPs.
若存cmg系统中的声码器单板上运行这种改进的EVRC编解码算法，在支持相同数目的EVRC编解码器的情况下，能为系统节省一定的DSP数量。
If this codec is utilized on the vocoder board in the CMG system, with the same number of EVRC sustained, we can reduce quite a few DSPs.
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