一种基于匹配跟踪的谐波和独立谱线正弦模型实现方案

该文基于匹配跟踪的谐波和独立谱线正弦模型提出了一种用于参数音频编码的实现方案,输入音频信号的正弦成分由谐波联合独立谱线共同表示,分析合成过程采用50%叠接相加(OLA)消除帧间不连续,匹配跟踪(MP)算法在频域提取模型参数(幅度、频率和相位)大大降低运算复杂度。谐波谱线基频由谐波积谱法(SHS)获得,各次谐波频点确切位置借助MP迭代过程推出,并进行二次曲线拟合,对应的谐波幅度采用LPC谱包络近似。独立谱线的提取有效弥补了谐波提取不足。实验证明该文提出的正弦模型实现方案可以很好地表示出音频信号中的平稳成分,并对低比特率的参数音频编码有一定的借鉴性。     

Comparison of ISFs and LSFs in Speech/Music Discrimination System

The immittance spectral frequencies （ISFs） is proposed as a new set of classification features and compared with the linear spectral frequencies （LSFs） applied in a frame-level wideband speech/music discrimination system. These two sets of features can be shared by the classifier and coding module to reduce the total computational complexity, making our classification system suitable for multi-mode audio coding applications. A performance assessment and comparison of the features are made. The experiment results show that the ISFs and LSFs have similar good performance when using full covariance matrices in classification models and the ISFs perform slightly better when using diagonal matrices. Their statistical differences for speech and music signals are also revealed.     

基于OPNET仿真平台的AODV协议的性能研究

AODV(Ad hoc On-demand Distance Vector Routing)路由协议是面向移动Adhoc网络的,在介绍了移动Ad hoc网络的特征、现有路由协议和AODV路由协议的发展过程之后,重点分析了AODV协议的特征、路由建立、维护过程,并基于OPNET平台仿真了AODV协议的性能(包括路由发现时间、协议效率、平均跳数、数据吞吐量),总结了目前AODV协议存在的问题及其改进的主要方法,最后对AODV协议改进的前景进行了预测。     

适用于机舱环境的抗噪语音指令识别方案

基于飞行器语音指令控制的应用环境,提出了一种高抗噪性的非特定人连接词语音识别方案。针对机舱噪声的频谱特性,采用了谱减法、倒谱均值归一化以及噪声模型适应技术。仿真测试表明,该方案的平均识别率在信噪比为0 dB的条件下达到92%以上。     

Mapping methods for output-based objective speech quality assessment using data mining

Objective speech quality is difficult to be measured without the input reference speech. Mapping methods using data mining are investigated and designed to improve the output-based speech quality assessment algorithm. The degraded speech is firstly separated into three classes （unvoiced, voiced and silence）, and then the consistency measurement between the degraded speech signal and the pre-trained reference model for each class is calculated and mapped to an objective speech quality score using data mining. Fuzzy Gaussian mixture model （GMM） is used to generate the artificial reference model trained on perceptual linear predictive （PLP） features. The mean opinion score （MOS） mapping methods including multivariate non-linear regression （MNLR）, fuzzy neural network （FNN） and support vector regression （SVR） are designed and compared with the standard ITU-T P.563 method. Experimental results show that the assessment methods with data mining perform better than ITU-T P.563. Moreover, FNN and SVR are more efficient than MNLR, and FNN performs best with 14.50% increase in the correlation coefficient and 32.76% decrease in the root-mean-square MOS error.     

基于语音识别的汉语发音教学系统*

综述了基于语音识别的汉语发音教学系统,论述了该系统的思想框架及目前的研究进展,对其中的关键技术与原理进行了详细讨论,并就汉语发音教学系统提出了一些思路和想法。     

通信网络扩展的一种优化方法

讨论了目前可靠性评估的常用方法和相应的测度指标,指出了当前可靠性评估技术的缺点和不足。针对以添加链路的形式扩展通信网络的过程,提出了一种基于生成树的通信网络扩展的优化方法,并且利用这种方法进行了某种网络拓扑结构扩展的仿真计算,与现存的其它方法的比较表明,这种基于生成树的优化方法设计合理、计算快速,可以有效地应用于实际的通信网络扩展的优化过程中。     

Non-Intrusive Objective Speech Quality Measurement Based on Fuzzy GMM and SVR for Narrowband Speech

Based on fuzzy Gaussian mixture model (FGMM) and support vector regression (SVR),an improved version of non-intrusive objective measurement for assessing quality of output speech without inputting clean speech is proposed for narrowband speech.Its perceptual linear predictive (PLP) features extracted from clean speech and clustered by FGMM are used as an artificial reference model.Input speech is separated into three classes,for each a consistency parameter between each feature pair from test speech signals...     

Variable Rate Characteristic Waveform Interpolation Speech Coder Based on Phonetic Classification

A variable-bit-rate characteristic waveform interpolation （VBR-CWI） speech codec with about 1.8 kbit/s average bit rate which integrates phonetic classification into characteristic waveform （CW） decomposition is proposed. Each input frame is classified into one of 4 phonetic classes. Non-speech frames are represented with Bark-band noise model. The extracted CWs become rapidly evolving waveforms （REWs） or slowly evolving waveforms （SEWs） in the cases of unvoiced or stationary voiced frames respectively, while mixed voiced frames use the same CW decomposition as that in the conventional CWI. Experimental results show that the proposed codec can eliminate most buzzy and noisy artifacts existing in the fixed-bit-rate characteristic waveform interpolation （FBR-CWI） speech codec, the average bit rate can be much lower, and its reconstructed speech quality is much better than FS 1 016 CELP at 4.8 kbit/s and similar to G. 723.1 ACELP at 5.3 kbit/s.