介绍了自适应回声消除器的基本结构及各个功能模块，例如：变阶数自适应滤波器，语音状态检测器，延迟分析器和NL MS算法控制器。
The structure of AEC as well as its composing modules, are introduced such as the speech detector, the adaptive filter, the time delay estimator and the NLMS algorithm controller.
形态学滤波器是一种新的非线性语音增强算法。
Morphological filter is a kind of new nonlinear speech enhancement algorithm.
该波束形成器首先让信号经过维纳滤波器消除非相干噪声，而后经过经典的线性约束最小方差处理，最终达到较好的语音增强效果。
This kind of beamformer can eliminate incoherent noise when signals pass Wiener filter, then achieve a better speech-enhance effect by linear constraints minimum variance beamforming.
将TVAR模型的信号和反射系数矢量增广为状态矢量后，应用高斯粒子滤波器(GPF)估计TVAR的模型参数，构造了语音增强算法。
When TVAR model signal and reflection coefficients were extended to state vector, Gaussian Particle Filter (GPF) was applied to estimate parameters of TVAR model.
本文应用自适应滤波理论对煤矿井下架线电机车载波电话的语音增强问题进行研究。
This article applies from the adaptive-filtering theory on proceeding the research of the Speech Enhancement problem in the carrier telephone of the electric locomotive down the coal mine well.
利用此特性对信号进行小波域滤波，可从加噪的语音中提取人耳所能接受的频率成份，是一种简单有效的语音去噪算法。
Filtering signal in the wavelet domain, by the frequency components acceptable for ear from the original speech, is a simple and effective speech denoising method.
用该码本对声门波加以规范和约束，使迭代维纳滤波过程中增强语音的激励声门波处干干净语音有效激励声源模式空间内。
We imposes constraints on the glottal wave of enhanced speech during iterative Wiener filtering according as the glottal codebook to ensure glottal wave laid in the clean exciting glottal space.
本文利用大规模集成电路一开关电容滤波器SCF作为语音特征提取的基本组件，以微机为控制中心，设计了时分复用动态语谱分析系统。
This paper describes using large scale integrated circuit—switch capacitatice filter (SCF) as basic components, taking out the characteristic of speech and making up microcomputer into control centre.
数字滤波在语音和图像处理、HDTV、模式识别、谱分析等应用中经常用到。
It is useful of Digital filter in voice processing, image processing, HDTV, Pattern Recognition and spectrum analysis.
比较滤波前后语音信号的波形及频谱。
Compare filter before and after speech signal waveform and spectrum.
基于MATLAB语言编写，完成对语音信号的采集、频谱分析及滤波。
Based on MATLAB language, complete speech signal spectrum analysis and collecting and filtering.
与此同时，结合数字信号处理知识，分别用谐波修正和时变数字滤波器的方法，完成基于骨导信号的语音重构。
At the same time, based on the knowledge of digital signal processing, reconstruction by harmonic correction and time variant digital filter was proposed.
为了更好地提取说话人的特征，对语音进行滤波的预处理。
In order even better to extract feature parameters of speakers, speech should be filtered as a pre-processing process.
大多数实用的语音增强系统均是由精确的噪声估计器和良好的滤波器来实现对带噪语音进行去噪。
Most practical speech enhancement systems composed of an accuracy noise estimator and good filter to reduce the noise.
提出了一种基于RASTA滤波技术的多维语音特征和支持向量机分类的VAD算法，适用于低信噪比情况下的话音检测。
VAD algorithm based on RASTA-filter multi-dimensional speech feature and Support Vector Machine is presented. It applies to the speech detection under the low SNR conditions.
利用CTRANC抑制干扰信号的特性及语音信号的短时稳定性，借助最优控制相关理论，得到了新的语音分离方法及其自适应滤波迭代步长的计算公式。
Based on the CTRANC characteristic of suppressing the interfering signal and stability in short-term of speech signal, an adaptive speech separation algorithm based on the CTRANC system comes out.
分数延迟滤波器广泛用于通信，语音处理，回声消除等。
Fractional delay filters are applied to a wide range such as communications, speech processing, echo cancellation, etc.
针对某大型车辆专用车内通话系统设计中的语音增强问题，对一种基于自适应滤波的语音增强算法进行了较为深入的研究。
An algorithm based on adaptive filter is introduced to improve SNR in the design of a special communication system of large vehicle.
设计了一种可用于语音信号处理的CODEC芯片，讨论了滤波器组的多级实现。
This paper designs a CODEC chip of audio signal processing, and the multi-level filter is discussed.
本文介绍了一种利用DSP构建的基于自适应滤波算法的语音信号增强处理系统。
In this paper, a system of speech enhancement using DSP is discussed based on self-adaptive algorithm.
在感知加权处理中充分利用了人耳的掩蔽效应，设计了感知加权滤波器，对谱减法增强后的语音进行滤波，进一步消除残余噪声。
And in perceptual weighting stage, auditory masking effect is used. A perceptual weighting filter is designed to filter the enhanced speech by spectral subtraction so as to remove residual noise.
该算法重点研究经典LPC分析后基音激励方向向下的语音，对这种浊音LPC残差进行后滤波以取代预增强的方法使其逼近语音激励。
Firstly we analyze the speech with reversed polarity and present the idea of post-filtering of the LPC residual of these voiced speech frames.
听觉滤波器在理解听觉形成机制、听觉系统建模、语音压缩和语音识别等很多方面都有着很重要的应用。
Auditory filter plays an important role in understanding the mechanism of hearing, auditory modeling, speech compression and recognition.
在过去的几十年中，多速率数字滤波器组在许多领域已得到广泛的应用，如语音、图像处理和通信领域等。
In the past decades, multirate digital filter Banks attract many attentions in various fields, such as speech, image and communication applications.
为克服以往滤波器组参数调整复杂问题和实现电子耳蜗语音处理的快速数字化计算，提出了将小波变换应用于电子耳蜗的语音处理。
A new method on the basis of wavelet transform (WT) was presented to easily adjust the parameters of filter bank and realize fast algorithm of speech processing for cochlear implants.
本文设计的自适应滤波器主要是应用在电视机中，结合电视机语音控制系统的实际应用给出了设计模型，完成了模型的硬件化实现。
A design model is demonstrated in connection with the real implementation of the system of audio controlling in TV set, and this thesis completes implementing the model on hardware.
本文设计的自适应滤波器主要是应用在电视机中，结合电视机语音控制系统的实际应用给出了设计模型，完成了模型的硬件化实现。
A design model is demonstrated in connection with the real implementation of the system of audio controlling in TV set, and this thesis completes implementing the model on hardware.
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