区分性训练在声纹密码中的新应用

在声纹密码任务中由于数据稀疏的问题难以实现区分性训练,本文以一种表征距离度量的特征矢量为基础提出新的声纹密码区分性系统框架,对正反例样本的新特征矢量实现了基于最小分类错误准则的区分性训练,将声纹密码从确认问题转化为二类分类问题。在自由说话风格的60人数据集上,声纹密码区分性系统与混合高斯模型-通用背景模型(Gaussian mixture model-universal background model,GMM-UBM)系统融合后等错误率为4.48%,相对GMM-UBM,动态时间规划(Dynamic time warping,DTW)基线系统性能分别提升了17.95%和59.68%。     

基于傅立叶变换的多维数据可视化分类

基于雷达图表示多维数据的原理,提出一种利用傅立叶描述子识别雷达图形的可视化数据分类新方法。该方法采用多元统计中的雷达图表示多维数据,不同模式类别的多维数据构成不同形状的雷达图形。在此基础上对雷达图的边界曲线进行傅立叶变换,计算傅立叶描述子作为雷达图的图形特征,并运用改进的概率神经网络进行识别。实验结果表明该方法具有较好的分类效果,分类精度比传统分类方法提高了约8.25%。     

基于HMM-UBM的声纹密码识别

声纹识别中,提出基于隐马尔可夫－通用背景模型的识别算法。针对声纹密码中每个人的注册语音数据量很少的问题,提出使用大量其他人数据先建立话者无关的声韵母隐马尔可夫模型作为通用背景模型,再根据最大后验概率准则,以通用背景模型为基础使用训练语音自适应获得说话人模型。该方法解决在声纹密码识别中训练数据不足的问题。在讯飞桌面数据库Ⅱ上,采用该算法的系统的等错误率为6。8%。     

基于梅尔频率倒谱系数与动态时间规整的安卓声纹解锁系统

安卓设备通常采用数字或图形密码解锁,但此类口令形式的密码安全性不高,而且部分安卓版本存在锁屏绕过漏洞的问题。为此,设计一种利用用户声纹特征的安卓解锁系统。采用梅尔频率倒谱系数提取声纹特征,使用动态时间规整算法进行文本相关的声纹模式匹配,并结合安卓NDK技术实现快速声纹识别。实验结果表明,该系统具有较高的解锁成功率和较快的解锁速度,相比数字或图形解锁,声纹解锁安全性更高,用户体验更好。     

基于Log似然比的特征选择算法

针对基于向量空间模型文本分类系统中特征选择算法存在的问题,提出一种基于Log似然比的特征选择算法,引进Log似然比统计量,在考虑稀有事件对分类结果产生正面影响的同时,较好地控制其对分类产生的负面影响。采用KNN分类方法,将Log似然比特征选择算法与典型特征算法进行比较,实验结果表明,该算法能够获得良好的性能。     

多维数据特征融合的用户情绪识别

针对目前基于智能手机的情绪识别研究中所用数据较为单一,不能全面反应用户行为模式,进而不能真实反应用户情绪这一问题展开研究,基于智能手机从多个维度全面收集反应用户日常行为的细粒度感知数据,采用多维数据特征融合方法,利用支持向量机(support vector machine,SVM)、随机森林(random forest)等6种分类方法,基于离散情绪模型和环状情绪模型两种情绪分类模型,对12名志愿者的混合数据和个人数据分别进行情绪识别,并进行了对比实验。实验结果表明,该全面反应用户行为的多维数据特征融合方法能够很好地对用户的情绪进行识别,其中使用个人数据进行情绪识别的准确率最高可达到79.78%,而且环状情感模型分类结果明显优于离散分类模型。     

一种新的鲁棒声纹特征提取与融合方法

为提高说话人确认系统在噪声环境下的鲁棒性,在利用听觉外周模型改进Mel频率倒谱系数(Mel FrequencyCepstral Coefficient,MFCC)的基础上,结合感知线性预测系数(Perceptual Linear Predictive Coefficient,PLPC),以类间区分度为依据,在特征域对两种声纹特征进行融合,提出一种新的声纹特征提取方法,并对基于该特征的说话人确认系统的噪声鲁棒性进行研究。针对不同信噪比的语音信号进行了融合特征与原始特征的对比实验,结果表明,融合特征在模拟餐厅噪声环境中的错误率更低,较MFCC与PLPC分别降低了2.2%和3.1%,说话人确认系统在噪声中的鲁棒性得到提升。     

基于梯度提升决策模型的空间占用检测研究

随着绿色建筑和绿色生态城区经济激励机制基本形成,面对大量多维空间占用数据,“大数据绿色建筑”节能体系应运而生。然而大量多维的建筑数据却没有被充分利用,且传统空间占用检测模型分类精度还不够准确,模型时间复杂度较高。利用UCI占用检测数据集,在原始数据集上加入时间戳,使模型分类精度均获得提高,同时利用MCMR（最大相关最小冗余）方法进行特征选择,通过随机森林作为分类器验证分类效果,获取最优特征子集。且利用选取的特征子集构建占用检测模型,其中XGBoost模型与随机森林模型（RF）进行比对,分类精度较高,且时间复杂度更低。     

基于Transformer的有载分接开关故障诊断研究

针对传统网络捕捉有载分接开关声纹特征之间联系不充分导致故障诊断准确率低的问题,提出了基于Transformer神经网络的有载分接开关故障诊断方法。首先采用梅尔频率倒谱系数提取有载分接开关声纹特征,以降低有载分接开关声纹样本的数据维度。然后利用Transformer充分捕捉声纹特征之间的联系并实现有载分接开关故障诊断。实验结果表明,基于Transformer对有载分接开关传动轴松动、触头磨损、卡涩和连挡故障诊断的准确率高达97.5%,并一定程度缩短了诊断的时间。     

基于星型网模型的安全漏洞分类

在进行入侵检测技术的研究中,安全漏洞分类是一个重要和必需的过程。然而,现有的漏洞分类在不同程度上都存在一些弊端。为了使分类更加科学、有用,该文提出了一种基于星型网模型的安全漏洞分类。该分类利用星型网的特点,将所有漏洞构造成一个7维数据空间,每一维都有具体的粒度划分,可以利用多维数据模型上的数据挖掘对漏洞进行较全面的多维度的数据分析与知识发现。     

基于韵律特征的SVM说话人确认

提出了一种基于韵律特征和SVM的文本无关说话人确认系统。采用小波分析方法,从语音信号的MFCC、F0和能量轨迹中提取出超音段韵律特征,通过实验研究三者的韵律特征在特征层的最佳互补融合,得到信号的韵律特征PMFCCFE,用韵律特征的GMM均值超矢量作为参数训练目标话者的SVM模型,以更有效地区分目标话者和冒认话者。在NIST06 8side-1side数据库的实验表明,以短时倒谱参数的GMM-UBM系统为基准,超音段韵律特征的GMM-SVM系统的EER相对下降了57.9%,MinDCF相对下降了41.4%。     

Text-independent speaker verification using ant colony optimization-based selected features

With the growing trend toward remote security verification procedures for telephone banking, biometric security measures and similar applications, automatic speaker verification (ASV) has received a lot of attention in recent years. The complexity of ASV system and its verification time depends on the number of feature vectors, their dimensionality, the complexity of the speaker models and the number of speakers. In this paper, we concentrate on optimizing dimensionality of feature space by selecting relevant features. At present there are several methods for feature selection in ASV systems. To improve performance of ASV system we present another method that is based on ant colony optimization (ACO) algorithm. After feature reduction phase, feature vectors are applied to a Gaussian mixture model universal background model (GMM-UBM) which is a text-independent speaker verification model. The performance of proposed algorithm is compared to the performance of genetic algorithm on the task of feature selection in TIMIT corpora. The results of experiments indicate that with the optimized feature set, the performance of the ASV system is improved. Moreover, the speed of verification is significantly increased since by use of ACO, number of features is reduced over 80% which consequently decrease the complexity of our ASV system.     

Robust speaker recognition in cross-channel condition based on Gaussian mixture model

One of the most difficult challenges for speaker recognition is dealing with channel variability. In this paper, several new cross-channel compensation techniques are introduced for a Gaussian mixture model—universal background model (GMM-UBM) speaker verification system. These new techniques include wideband noise reduction, echo cancellation, a simplified feature-domain latent factor analysis (LFA) and data-driven score normalization. A novel dynamic Gaussian selection algorithm is developed to reduce the feature compensation time by more than 60% without any performance loss. The performance of different techniques across varying channel train/test conditions are presented and discussed, finding that speech enhancement, which used to be neglected for telephone speech, is essential for cross-channel tasks, and the channel compensation techniques developed for telephone channel speech also perform effectively. The per microphone performance analysis further shows that speech enhancement can boost the effects of other techniques greatly, especially on channels with larger signal-to-noise ratio (SNR) variance. All results are presented on NIST SRE 2006 and 2008 data, showing a promising performance gain compared to the baseline. The developed system is also compared with other state-of-the-art speaker verification systems. The result shows that the developed system can obtain comparable or even better performance but consumes much less CPU time, making it more suitable for practical use.     

基于对话语音的与文本无关的说话人确认系统的研究

本文建立了一个基于对话语音的与文本无关的说话人确认系统,它和传统的与文本无关的说话人确认系统的关键不同在于,训练及测试语音不再只包含一个人而都是对话语音,因此需要分割出属于不同说话人的语音段,以建立说话人模型和实现最终判决。文中详细介绍了高斯混合模型-背景模型(GMM-UBM)这种说话人确认系统的框架,重点讨论了基于GLR(Generalized Likelihood Ratio)距离测度的无监督语音分割算法。最终阐述的输出评分的规整方法即ZNORM(Zero Normalization)和持续时间修正,可以使确认系统的性能提高近10%。     

Speaker verification using excitation source information

In this work we develop a speaker recognition system based on the excitation source information and demonstrate its significance by comparing with the vocal tract information based system. The speaker-specific excitation information is extracted by the subsegmental, segmental and suprasegmental processing of the LP residual. The speaker-specific information from each level is modeled independently using Gaussian mixture modeling—universal background model (GMM-UBM) modeling and then combined at the score level. The significance of the proposed speaker recognition system is demonstrated by conducting speaker verification experiments on the NIST-03 database. Two different tests, namely, Clean test and Noisy test are conducted. In case of Clean test, the test speech signal is used as it is for verification. In case of Noisy test, the test speech is corrupted by factory noise (9 dB) and then used for verification. Even though for Clean test case, the proposed source based speaker recognition system still provides relatively poor performance than the vocal tract information, its performance is better for Noisy test case. Finally, for both clean and noisy cases, by providing different and robust speaker-specific evidences, the proposed system helps the vocal tract system to further improve the overall performance.     

基于模型距离和支持向量机的说话人确认

针对采用支持向量机的说话人的确认问题,提出采用背景模型、说话人模型、测试语句模型间距离和夹角作为支持向量机的特征矢量,同时将组特征矢量与广义线性判别式序列核函数的参数相拼接,能够取得相对于基线的混合高斯模型算法更高的识别率.在2004年NIST评测数据库上,采用推荐算法的系统等错误率比基线的混合高斯-背景模型系统低16%.对说话人识别取得一定进展.     

Fast and Accurate 3D Face Recognition

In this paper we present a new robust approach for 3D face registration to an intrinsic coordinate system of the face. The intrinsic coordinate system is defined by the vertical symmetry plane through the nose, the tip of the nose and the slope of the bridge of the nose. In addition, we propose a 3D face classifier based on the fusion of many dependent region classifiers for overlapping face regions. The region classifiers use PCA-LDA for feature extraction and the likelihood ratio as a matching score. Fusion is realised using straightforward majority voting for the identification scenario. For verification, a voting approach is used as well and the decision is defined by comparing the number of votes to a threshold. Using the proposed registration method combined with a classifier consisting of 60 fused region classifiers we obtain a 99.0% identification rate on the all vs first identification test of the FRGC v2 data. A verification rate of 94.6% at FAR=0.1% was obtained for the all vs all verification test on the FRGC v2 data using fusion of 120 region classifiers. The first is the highest reported performance and the second is in the top-5 of best performing systems on these tests. In addition, our approach is much faster than other methods, taking only 2.5 seconds per image for registration and less than 0.1 ms per comparison. Because we apply feature extraction using PCA and LDA, the resulting template size is also very small: 6 kB for 60 region classifiers.     

采用模型间距离夹角的说话人确认测试算法

提出在与文本无关说话人确认中采用模型间马氏(Mahalanobis)距离的夹角作为测试算法,在混合高斯模型(Gaussian ixture Model)的情况下,采用这种算法在保持识别率与传统的对数似然度算法相近的前提下,可以大大降低运算量,对于说话人确认或识别的实时实现有很大的帮助.另外,推荐的算法与传统的对数似然度算法的结果可以融合,可以将说话人确认的等错误率降低12～15%.     

A novel scores fusion approach applied on speaker verification under noisy environments

To improve the speaker verification system in adverse conditions, a novel score fusion approach using adaptive method, based on a prior Equal Error Rate (EER), is presented in this paper. Currently, the most commonly used methods are the mean, product, minimum, maximum, or the weighted sum of scores. Our method introduces the MLP network which approximates the estimated scores under noisy conditions, to those of the ideal estimated in clean environments and gives the optimally weighted parameters, to be added in the adaptive weights used for weighting sum of scores. This method is assessed by using the NIST 2000 corpus and different feature extraction methods. Noisy conditions are created using NOISEX-92. In severely degraded conditions, the results show that the speaker verification process using our proposed score fusion approach applied to the GMM-UBM and GMM-SVM based systems, achieves better performances in terms of EER reduction than each system used alone.