一种改进的模糊C-均值聚类算法在说话人识别中的应用

提出了一种将改进的FCM聚类算法与矢量量化相结合的说话人识别的方法。先从语音信号中提取待识别的特征矢量集,再利用矢量量化来设计码本,最后用改进的算法对待识别语音进行辩识。该算法解决了FCM算法对初始值敏感、易陷入局部最优的问题。所使用的特征参数较少,计算比较简单,但识别率较高,且具有较好的鲁棒性。     

高效的模糊聚类初始码书生成算法

码书设计在矢量量化中至关重要,而多数码书设计算法都是基于初始码书的.从经典的LBG算法的缺陷出发,提出一种基于模糊聚类的高效初始码书生成算法,通过将初始码书的码矢在输入矢量空间中很好地散开,并尽可能占据输入概率密度较大的区域,从而使之后的LBG算法避免陷入局部最优,设计出的码书性能更好,更加接近全局最优,同时加快了收敛速度,减少了迭代次数.将该算法应用于图像编码的实验中,结果表明:该算法能够从效率和质量两方面有效地提高矢量量化的性能.     

基于可区分性加权的模糊核说话人识别

 针对训练和识别语音数据较少的情况,本文提出了一种新的说话人识别算法.通过核映射,在高维特征空间对说话人的语音特征进行模糊矢量量化.为了增加说话人之间的可区分性,提出了一种基于高维特征空间的码字矢量的权值分配方法,对具有较强区分性的码字矢量分配较大的权值,并将产生的权值和说话人的码书一起形成说话人数据库.识别时,提出一种模糊核加权最近邻近分类器,在高维特征空间中对说话人进行匹配.实验表明,该算法在训练语音少于8s,识别语音为1s时,能够得到较好的识别结果.     

模糊矢量量化在语音情感识别中的应用

提出了一种将改进的模糊C均值聚类算法与矢量量化相结合的语音情感识别方法,实现了对4种情感的识别:高兴、生气、悲伤和惊奇.首先提取情感语句全局结构和时序结构特征参数并进行性别规整.再利用改进后的模糊矢量量化方法来设计码本,最后对待识别语音进行辩识.该算法不但解决了模糊C均值算法对初始值敏感、易陷入局部最优的问题,而且性别规整改善了特征参数的有效性,使识别率得以进一步提高.实验结果表明该算法能够有效改善识别率.     

基于聚类统计与文本无关的说话人识别研究

从语音信号特征矢量的空间映射出发,在二元分裂算法的基础上提出了一种裂合并的聚类算法,并用于与本文无关的说话人识别研究,初步建立了基于聚类统计的开放系统,该系统用说话人语音信号在特征空间的分布中心建立参考模板,用聚类统计中心代替待识语音段的特征矢量进行了模式匹配计算,系统规模的越大,节省的计算量就越多。在小规模说话人辨认系统的实践研究中,研究了特征矢量的加权,语音段的时长以及a因子的选择对系统性能的影响。     

基于改进Kohonen网络训练算法的说话人识别系统

与文本无关的说话人识别方法是当前说话人识别技术的研究重点.本文将自组织特征映射(SOFM)网络用于矢量量化中码书的形成,改进了训练算法,并基于改进算法进行了与文本无关的说话人识别.经实验结果证明,本文的方法改善了码本的性能,提高了说话人识别的识别率.     

应用矢量量优的孤立词汉语语音识别

本文提出了一种识别孤立词汉语语音的新方法,提取线性预测系数作为语音特征,利用矢量量化的聚类特性压缩数据,用多段码书作为语音样板,用最小失真法进行识别。     

一种快速模糊矢量量化图像编码算法

本文在学习矢量量化和模糊矢量量化算法的基础上,设计了一种新的训练矢量超球体收缩方案和码书学习公式,提出了一种快速模糊矢量量化算法。该算法具有对初始码书选取信赖性小,不会陷入局部最小和运算最小的优点。实验表明,ＦＦＶＱ设计的图像码书性能与ＦＶＡ算法相比,训练时间大大缩短,峰值信噪比也有改善。     

基于BWT和FVQ的极低比特率图像编码算法

该文提出了一种基于双正交小波变换(BWT)和模糊矢量量化(FVQ)的极低比特率图像编码算法。该算法通过构造符合图像小波变换系数特征的跨频带矢量,充分利用了不同频带小波系数之间的相关性,有效地提高了图像的编码效率和重构质量。该算法采用非线性插补矢量量化(NLIVQ)的思想,从大维数矢量中提取小维数的特征矢量,并提出了一种新的模糊矢量量化方法一渐进构造模糊聚类(PCFC)算法用于特征矢量的量化,从而大大提高了矢量量化的速度和码书质量。实验结果证明,该算法在比特率为0．172bpp的条件下仍能获得PSNR＞30dB的高质量重构图像。     

新型混合矢量码书快速匹配算法

矢量量化码书快速匹配算法是矢量量化技术实时应用的关键所在。从原理上看，提高矢量码书匹配速度的方法可以分为两类：一类是通过建立某种“剔除”条件来避免不必要的输入矢量和码书矢量之间欧氏距离计算“软”算法，另外一类是通过找表（LUT）技术来代替欧氏距离计算中乘法运算的“硬”方法，文章利用这两类方法的互补性，提出了一种“软”、“硬”方法相结合的矢量码书快速匹配算法。实验表明，该算法具有非常好的实时性能，能够方便地应用于加快数据压缩矢量量化的码书匹配过程以及模式识别中SOM型神经网络的识别过程。     

Novel fuzzy reinforced learning vector quantisation algorithm and its application in image compression

A new approach to the design of optimised codebooks using vector quantisation (VQ) is presented. A strategy of reinforced learning (RL) is proposed which exploits the advantages offered by fuzzy clustering algorithms, competitive learning and knowledge of training vector and codevector configurations. Results are compared with the performance of the generalised Lloyd algorithm (GLA) and the fuzzy K-means (FKM) algorithm. It has been found that the proposed algorithm, fuzzy reinforced learning vector quantisation (FRLVQ), yields an improved quality of codebook design in an image compression application when FRLVQ is used as a pre-process. The investigations have also indicated that RL is insensitive to the selection of both the initial codebook and a learning rate control parameter, which is the only additional parameter introduced by RL from the standard FKM     

基于FVQ/HMM的无教师说话人自适应

本文提出了一种新的语音识别方法,它综合了VQ、HMM和无教师说话人自适应算法的优点,在每个状态通过用矢量量化误差值取代传统HMM的输出概率值来建立FVQ/HMM,同时采用基于模糊矢量量化的无教师自适应算法,来改变FVQ/HMM的各状态的码字,从而实现对未知说话人的码本适应.本文通过非特定人汉语数码(孤立和连续数码)语音识别实验,把该新的组合方法同基于CHMM的自适应和识别方法进行了比较,实验结果表明该方法的自适应和识别效果优于基于CHMM的方法.     

基于动词相似度的特征分类方法

本文提出了一种新的说话人特征分类方法,基于计算动词相似度理论,建立距离和趋势的评价模型,通过计算特征向量与k-means算法聚类所得的聚类中心的相似度矩阵,将说话人个性特征从MFCC特征域映射到说话人相似度属性空间中,形成新的特征向量集,这样,每个说话人的特征向量将被聚为在距离和变化趋势上最具相似性的k分类。之后,利用GMM模型在属性空间内进行联合概率分析、匹配,建立新的说话人识别系统。本文采用标准TIMIT语音库与NIST语音库在该识别系统中进行一系列实验,结果表明,该基于新的优化特征分类的识别系统,对比传统的说话人识别系统,在等错误率上有很好的提高。     

高阶异构数据模糊联合聚类算法

为了更有效地分析聚簇重叠部分高阶异构数据的聚簇结果,提出了一种高阶异构数据模糊联合聚类（HFCC）算法,该算法最小化每个特征空间中对象与聚簇中心的加权距离。推导出对象隶属度和特征权重的迭代更新公式,设计出聚类过程的迭代算法,并且从理论上证明了该迭代算法的收敛性。另外,通过泛化XB指标,提出适用于评估高阶异构数据聚类质量的指标GXB,用于判断聚簇数目。实验表明,HFCC算法能够有效探测数据内部隐藏的重叠聚簇结构,并且HFCC算法聚类效果明显优于5种有代表性的硬划分算法,此外GXB指标能够有效判定高阶异构数据的聚簇数目。     

Linearly constrained generalized Lloyd algorithm for reducedcodebook vector quantization

As linearly constrained vector quantization (LCVQ) is efficient for block-based compression of images that require low complexity decompression, it is a “de facto” standard for three-dimensional (3-D) graphics cards that use texture compression. Motivated by the lack of an efficient algorithm for designing LCVQ codebooks, the generalized Lloyd (1982) algorithm (GLA) for vector quantizer (VQ) codebook improvement and codebook design is extended to a new linearly constrained generalized Lloyd algorithm (LCGLA). This LCGLA improves VQ codebooks that are formed as linear combinations of a reduced set of base codewords. As such, it may find application wherever linearly constrained nearest neighbor (NN) techniques are used, that is, in a wide variety of signal compression and pattern recognition applications that require or assume distributions that are locally linearly constrained. In addition, several examples of linearly constrained codebooks that possess desirable properties such as good sphere packing, low-complexity implementation, fine resolution, and guaranteed convergence are presented. Fast NN search algorithms are discussed. A suggested initialization procedure halves iterations to convergence when, to reduce encoding complexity, the encoder considers the improvement of only a single codebook for each block. Experimental results for image compression show that LCGLA iterations significantly improve the PSNR of standard high-quality lossy 6:1 LCVQ compressed images     

模糊强化学习型的图像矢量量化算法

本文给出了一种新的图像矢量量化码书的优化设计方法.传统矢量量化方法只考虑了码字与训练矢量之间的吸引影响,所以约束了最优解的寻解空间.本文提出了一种新的学习机理--模糊强化学习机制,该机制在传统的吸引因子基础上,引入新的排斥因子,极大地释放了吸引因子对最优解的寻解空间的约束.新的模糊强化学习机制没有采用引入随机扰动的方法来避免陷入局部最优码书,而是通过吸引因子和排斥因子的合力作用,较准确地确定了每个码字的最佳移动方向,从而使整体码书向全局最优解靠近.实验结果表明,基于模糊强化学习机制的矢量量化算法始终稳定地取得显著优于模糊K-means算法的性能,较好地解决了矢量量化中的码书设计容易陷入局部极小和初始码书影响优化结果的问题.     

基于分段模糊聚类算法的VQ-HMM语音识别模型参数估计

提出了一种新的语音识别模型VQ－HMM。它综合VQ和HMM的优点，在每个状态通过用矢量量化误差值取代传统HMM的输出概率值来建立模型，同时提出了基于分段模糊聚类算法的模型参数估计方法。文中给出了模型的实现方法，并通过实验证明了其在语音识别中具有很好的性能。     

Constrained-storage vector quantization with a universal codebook

Many image compression techniques require the quantization of multiple vector sources with significantly different distributions. With vector quantization (VQ), these sources are optimally quantized using separate codebooks, which may collectively require an enormous memory space. Since storage is limited in most applications, a convenient way to gracefully trade between performance and storage is needed. Earlier work addressed this problem by clustering the multiple sources into a small number of source groups, where each group shares a codebook. We propose a new solution based on a size-limited universal codebook that can be viewed as the union of overlapping source codebooks. This framework allows each source codebook to consist of any desired subset of the universal code vectors and provides greater design flexibility which improves the storage-constrained performance. A key feature of this approach is that no two sources need be encoded at the same rate. An additional advantage of the proposed method is its close relation to universal, adaptive, finite-state and classified quantization. Necessary conditions for optimality of the universal codebook and the extracted source codebooks are derived. An iterative design algorithm is introduced to obtain a solution satisfying these conditions. Possible applications of the proposed technique are enumerated, and its effectiveness is illustrated for coding of images using finite-state vector quantization, multistage vector quantization, and tree-structured vector quantization.     

Speaker identification based on adaptive discriminative vector quantisation

A novel adaptive discriminative vector quantisation technique for speaker identification (ADVQSI) is introduced. In the training mode of ADVQSI, for each speaker, the speech feature vector space is divided into a number of subspaces. The feature space segmentation is based on the difference between the probability distribution of the speech feature vectors from each speaker and that from all speakers in the speaker identification (SI) group. Then, an optimal discriminative weight, which represents the subspace's role in SI, is calculated for each subspace of each speaker by employing adaptive techniques. The largest template differences between speakers in the SI group are achieved by using optimal discriminative weights. In the testing mode of ADVQSI, discriminative weighted average vector quantisation (VQ) distortions are used for SI decisions. The performance of ADVQSI is analysed and tested experimentally. The experimental results confirm the performance improvement employing the proposed technique in comparison with existing VQ techniques for SI and recently reported discriminative VQ techniques for SI (DVQSI)