语音信号

结果表明，将汉语的音调变化信息加入到CIS语音信号处理方案中，可明显地提高汉语的识别能力。

Results showed that adding tonal information to the CIS speech processor would offer an improved intelligibility of spoken Chinese .

在深入研究码激励线性预测编码算法理论的基础上，提出了基于FS-1016标准实现数字语音信号编解码的技术方案，并开发了数字语音信号的编解码仿真系统。

Basing on the study of FS-1016 algorithm , a digital speech codec scheme in technic is presented and a simulation system is constructed for it .

基于MATLAB的语音信号采集和分析系统的可视化设计

Visual design of audio signal acquisition and analysis system based on the MATLAB

语音信号的修正AR模型

The Revised Auto - Regressive Model of Speech Signal

Matlab仿真结果表明三种方案的数据压缩率高，重建语音信号具有良好的清晰度和自然度。

MATLAB simulative results indicate that their data compression ratio is high and the reconstructed speech signals have good quality .

经过MATLAB仿真模拟以及DSP平台的测试，本设计初步实现了语音信号的加密解密过程。

The process of voice signal encryption and decryption is achieved through the examiner of MATLAB simulation and DSP platform .

IP电话将语音信号数字化后，再以数据封包（DataPacket）的形式在IP数据网络上进行实时传输。

IP phone is digitized as voice signal , then transmit as data packet ( Data Packet ) in the IP data networks in real time fashion .

语音信号ADM编码的改进

An Improvement on Speech Signal ADM Coding and Decoding

文中采用硬件方法对人工耳蜗语音信号处理的CIS算法加以实现。

CIS algorithm of cochlear implants ′ speech signal processing is achieved through hardware method .

这也是稳健（Robust）语音信号处理中的一个重要研究方向，对语音识别、语音增强等都有着非常积极的促进意义。

This is also a significant research direction in robust speech signal processing , and it has very active signification to speech recognition and speech enhancement .

语音信号MP稀疏分解快速算法及在语音识别中的初步应用

Research on Fast Algorithm for Speech Signal MP Sparse Decomposition and Its Application in Speech Recognition

IP电话是通过国际互联网络（Internet）来传输语音信号的通信设备，由于它具有节省通信带宽和通信费用等优点，因而备受关注。

IP phone is a communication equipment of transmitting speech signals over Internet . It has been paid more attention now because of the advantage of saving bandwith and money .

其中，针对相邻频点间存在的语音信号乱序问题，结合相关最大方法改进了频域mask聚类调序算法。

For the permutation problems between adjacent frequency bins , an improved mask clustering reordering algorithm combined with correlation maximum was proposed .

首先，本文通过对HELP算法的深入分析，根据语音信号谐波相关程度能反映浊音度强弱的性质，开发了一种基于最小均方误差准则的谐波相关浊音度参数提取方法。

First , after deeply investigating HELP model , a harmonic related voicing detection algorithm based on MSE criterion is developed , with the knowledge that voicing algorithm can be showed by degree of harmonic relation .

线性预测技术主要用于估计基本的语音信号的参数，目标是求得一组预测器系数和预测误差（LPCPE，LPCPredictionError）。

LPC technique is used to estimate the basic parameters of digital speech signals , which aims at obtaining a set of prediction coefficients and prediction error .

基于CEP和LPC谱提取语音信号基音周期的方法

Speech Signal Extraction Method of Pitch Based on CEP and LPC Spectrum

语音信号的加权mel倒谱分析

Weighted mel - cepstrum for speech analysis

AnnexB提出了一种静音压缩算法（VAD），它将语音信号分为话音信号和背景噪声信号。

Annex B introduce a voice activity decision ( VAD ) algorithm which class speech signal as voice signal and background noise signal .

结合CPU两个内核不同的体系结构，在DSP内核上连接A/D，D/A模块实现语音信号I/O通道以及实时实现语音编解码算法；

According to systems architecture , the A / D and D / A Interface are both connected to a DSP processor to realize speech signal I / O channels .

采用基于动态规划方法的动态时间归正技术DTW（DynamicTimewarping），可成功解决语音信号特征参数序列比较时时长不等的问题。

The dynamic time warping ( DTW ) technology based on the dynamic programming can solve the problem that the characteristic parameters of speech signal are of different lengths in each other .

应用EMD分解法并结合短时分析技术，处理语音信号，提出了三种特征提取算法。

The EMD , together with the short-time analysis , is used to analyze speech signals to extract three kinds of speaker features .

DSP（数字信号处理器）具有在单机器周期内完成乘加运算、单机器周期内多次访问存储器以及丰富的片上外设等特点，采用DSP进行语音信号处理代表未来语音信号处理的发展方向。

DSP ( digital sigal processor ) features Single-cycle multiply and accumulate ( MAC ), mutilple-accesss memory architecture , a wide variety of on-chip peripherals , which leads trend in digital speech processing field .

WI编码器将语音信号表示为渐变的特征波形（CW），并将其分解为慢渐变波形（SEW）与快渐变波形（REW），以分别进行量化。

In the WI coder , the speech signal is represented by an evolving characteristic waveform ( CW ) .

可以求出语音信号的LPC倒谱特征向量，该特征向量在语音信号分析中得到了广泛的应用。

Voice signal can be obtained by LPCCEP eigenvector , the eigenvector of the voice signal analysis has been widely used .

利用MATLAB软件和WINDOWS的录音机资源，把录取的声音信号转变为离散的数字信号，实现语音信号的采集和处理，并对上述研究做了大量的实验和仿真。

The transcribed voice signals are transformed into discrete digital signals with the matlab and recorder of windows to collect and deal with speech signals . During the procedure a lot of experiments and emulation are made .

VQ模型是一个用语音信号特征的分布中心描述说话人的个性特征而没有描述语音信号特征时序性的模型。

VQ model describes speaker characteristics with the speech feature centroids and it can 't represent the correlation in the speech feature frames .

深入分析线性预测在语音信号分析合成中的应用，实现基于Burg逆运算的语音信号的合成。

Secondly , the linear prediction can be applied to speech analysis and synthesis , with the compressing technique .

对各对等体语音信号的处理方面利用Windows、Visualc++的语音编程，使用了一种采集/发送乒乓机制和接收/播放乒乓机制来实现语音信号采集、发送、接收、播放的连续性。

Uses Windows , Visual C + + voice programming to dispose each peer 's voice signal , uses pick / transmit pingpong mechanism and the receive / broadcast pingpong mechanism realizes voice signal gathering , transmission , receiving , broadcast continuously .

依据互信息理论提出的互信息匹配识别模型MIM（MutualInformationMatching），能够有效地综合处理语音信号的统计分布特征与时变分布特征，并具有较强的鲁棒性。

The Mutual Information Matching model ( MIM ) was proposed for speaker recognition based on the mutual information theory . Both of statistical and time-variant features of speech signal can be processed effectively , robustly and synchronously in MIM .

本文由基于HHT方法提取的HF特征建立了一个说话人确认系统，所采用的全部语音信号数据均来自YOHOspeakerverification语音库。

This text set up a speaker verification system on the basis of HF characteristic that draw from HHT , all speech signal adopted come from YOHO speaker verification speech corpus .