基于HMM和聚类的英语语音识别混合算法

对于具有大量特征数据和复杂发音变化的英语语音,与单词相比,在隐马尔可夫模型(HMM)中存在更多问题,例如维特比算法的复杂度计算和高斯混合模型中的概率分布问题。为了实现基于HMM和聚类的独立于说话人的英语语音识别系统,提出了用于降低语音特征参数维数的分段均值算法、聚类交叉分组算法和HMM分组算法的组合形式。实验结果表明,与单个HMM模型相比,该算法不仅提高了英语语音的识别率近3%,而且提高系统的识别速度20.1%。     

Off-line handwritten word recognition using a hidden Markov modeltype stochastic network

Because of large variations involved in handwritten words, the recognition problem is very difficult. Hidden Markov models (HMM) have been widely and successfully used in speech processing and recognition. Recently HMM has also been used with some success in recognizing handwritten words with presegmented letters. In this paper, a complete scheme for totally unconstrained handwritten word recognition based on a single contextual hidden Markov model type stochastic network is presented. Our scheme includes a morphology and heuristics based segmentation algorithm, a training algorithm that can adapt itself with the changing dictionary, and a modified Viterbi algorithm which searches for the (l+1)th globally best path based on the previous l best paths. Detailed experiments are carried out and successful recognition results are reported     

An on-line adaptive neural network for speech recognition

In this paper, we present an on-line learning neural network model, Dynamic Recognition Neural Network (DRNN), for real-time speech recognition. The property of accumulative learning of the DRNN makes it very suitable for real-time speech recognition with on-line learning. A comparison between the DRNN and Hidden Markov Model (HMM) shows that the computational complexity of the former is lower than that of the latter in both training and recognition. Encouraging results are obtained when the DRNN is tested on a BUPT digit database (Mandarin) and on the on-line learning of twenty isolated English computer command words.     

Merge-Weighted Dynamic Time Warping for Speech Recognition

Obtaining training material for rarely used English words and common given names from countries where English is not spoken is di?cult due to excessive time, storage and cost factors. By considering pe...     

Robust speech recognition based on joint model and feature spaceoptimization of hidden Markov models

The hidden Markov model (HMM) inversion algorithm, based on either the gradient search or the Baum-Welch reestimation of input speech features, is proposed and applied to the robust speech recognition tasks under general types of mismatch conditions. This algorithm stems from the gradient-based inversion algorithm of an artificial neural network (ANN) by viewing an HMM as a special type of ANN. Given input speech features s, the forward training of an HMM finds the model parameters lambda subject to an optimization criterion. On the other hand, the inversion of an HMM finds speech features, s, subject to an optimization criterion with given model parameters lambda. The gradient-based HMM inversion and the Baum-Welch HMM inversion algorithms can be successfully integrated with the model space optimization techniques, such as the robust MINIMAX technique, to compensate the mismatch in the joint model and feature space. The joint space mismatch compensation technique achieves better performance than the single space, i.e. either the model space or the feature space alone, mismatch compensation techniques. It is also demonstrated that approximately 10-dB signal-to-noise ratio (SNR) gain is obtained in the low SNR environments when the joint model and feature space mismatch compensation technique is used.     

基于差分过零峰值幅度特征的抗噪语音识别

针对多数语音识别系统在噪音环境下性能急剧下降的问题,提出了一种新的语音识别特征提取方法。该方法是建立在听觉模型的基础上,通过组合语音信号和其差分信号的上升过零率获得频率信息,通过峰值检测和非线性幅度加权来获取强度信息,二者组合在一起,得到输出语音特征,再分别用BP神经网络和HMM进行训练和识别。仿真实现了不同信噪比下不依赖人的50词的语音识别,给出了识别的结果,证明了组合差分信息的过零与峰值幅度特征具有较强的抗噪声性能。     

基于HMM与RBF的混合语音识别新方法

提出了一种隐马尔可夫模型（ＨＭＭ）和径向基函数神经网络（ＲＢＦ）相结合的语音识别新方法。该方法首先利用ＨＭＭ生成最佳语音状态序列,然后用函数逼近技术产生对最佳状态序列进行时间规正,最后通过ＲＢＦ神经网络进行分类识别。理论和实验结果表明,该系统比ＨＭＭ具有更好的识别效果,特别对提高易混淆词的识别性能尤为显著。     

基于HMM与遗传神经网络的改进语音识别系统

为了解决语音信号中帧与帧之间的重叠,提高语音信号的自适应能力,本文提出基于隐马尔可夫(HMM)与遗传算法神经网络改进的语音识别系统.该改进方法主要利用小波神经网络对Mel频率倒谱系数(MFCC)进行训练,然后利用HMM对语音信号进行时序建模,计算出语音对HMM的输出概率的评分,结果作为遗传神经网络的输入,即得语音的分类识别信息.实验结果表明,改进的语音识别系统比单纯的HMM有更好的噪声鲁棒性,提高了语音识别系统的性能.     

一种基于LDAO的唇读特征提取算法

在语音与唇读识别应用中,传统的LDA(linear discriminant analysis)算法一般以音节、半音节、HMM状态等基元为类别进行数据分段,经线性判别分析后获得的特征投影方向与识别率不直接相关,影响了识别率。提出了一种新的基于LDAO(linear discriminant analysis based on object)的唇读特征提取算法,该算法以待识别对象为类别进行线性判别分析,在理论上保证了唇读特征矢量向最具判别能力的方向投影。基于唇读数据库的实验证明,该算法明显优于现有各种唇读特征提取算法,比DCT+LDA算法识别率提高了3%。     

基于动态时间规整和隐马尔可夫统一模型的无端点检测的汉语识别算法

根据汉语语音的特点，提出了一种无端点检测的语音识别算法。在识别过程中，该算法无需确定语音信号起止点位置，而是从寂静段开始，直接按帧提取特征（帧长２０ｍｓ，帧间重叠５０％），特征向量由１５阶倒谱系数和帧平均能量组成。在动态时间规整（ＤＴＷ）和隐马尔可夫（ＨＭＭ）统一模型（ＤＨＵＭ）中，引进寂静段自环，并用ＤＨＵＭ实现了该算法。对９９个相似汉语单字的识别实验表明：无端点检测的识别器正识率为９４．９５％，正识率下降很少，但不作端点检测却降低了算法的复杂程度。该算法中，若特征向量采用一种听觉模型特征，识别器具有更好的鲁棒性，识别率会略有提高。     

Comparing ANN to HMM in implementing limited Arabic vocabulary ASR systems

In this paper we investigated Artificial Neural Networks (ANN) based Automatic Speech Recognition (ASR) by using limited Arabic vocabulary corpora. These limited Arabic vocabulary subsets are digits and vowels carried by specific carrier words. In addition to this, Hidden Markov Model (HMM) based ASR systems are designed and compared to two ANN based systems, namely Multilayer Perceptron (MLP) and recurrent architectures, by using the same corpora. All systems are isolated word speech recognizers. The ANN based recognition system achieved 99.5% correct digit recognition. On the other hand, the HMM based recognition system achieved 98.1% correct digit recognition. With vowels carrier words, the MLP and recurrent ANN based recognition systems achieved 92.13% and 98.06, respectively, correct vowel recognition; but the HMM based recognition system achieved 91.6% correct vowel recognition.     

Efficient backward decoding of high-order hidden Markov models

The forward-backward search (FBS) algorithm [S. Austin, R. Schwartz, P. Placeway, The forward-backward search algorithm, in: Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing, 1991, pp. 697-700] has resulted in increases in speed of up to 40 in expensive time-synchronous beam searches in hidden Markov model (HMM) based speech recognition [R. Schwartz, S. Austin, Efficient, high-performance algorithms for N-best search, in: Proceedings of the Workshop on Speech and Natural Language, 1990, pp. 6-11; L. Nguyen, R. Schwartz, F. Kubala, P. Placeway, Search algorithms for software-only real-time recognition with very large vocabularies, in: Proceedings of the Workshop on Human Language Technology, 1993, pp. 91-95; A. Sixtus, S. Ortmanns, High-quality word graphs using forward-backward pruning, in: Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing, 1999, pp. 593-596]. This is typically achieved by using a simplified forward search to decrease computation in the following detailed backward search. FBS implicitly assumes that forward and backward searches of HMMs are computationally equivalent. In this paper we present experimental results, obtained on the CallFriend database, that show that this assumption is incorrect for conventional high-order HMMs. Therefore, any improvement in computational efficiency that is gained by using conventional low-order HMMs in the simplified backward search of FBS is lost.This problem is solved by presenting a new definition of HMMs termed a right-context HMM, which is equivalent to conventional HMMs. We show that the computational expense of backward Viterbi-beam decoding right-context HMMs is similar to that of forward decoding conventional HMMs. Though not the subject of this paper, this allows us to more efficiently decode high-order HMMs, by capitalising on the improvements in computational efficiency that is obtained by using the FBS algorithm.     

基于HTK的语音识别的并行化研究与实现

详细地分析了语音识别的过程,给出了相应的算法描述,并分析了语音识别并行化的可能性。将并行计算的思想应用于语音识别的算法中,使用多线程技术,并引入避免竞争条件的机制,在多核计算机上并行地计算HMM模型节点的似然率,从而得到语音识别的并行化算法。分析了该并行化算法的性能,同时在语音识别工具包HTK 3.4上实现了这种并行化算法。基于WSJ0语料库的实验结果表明该并行化算法在不影响识别结果的前提下能够有效地提高语音识别的实时性能。     

隐马尔可夫模型的一种有区分力的反向传播训练方法

研究隐马尔可夫模型(HMM)的一种有区分力的训练方法.在多层前向神经网络的 框架中实现了HMM的前向概率计算.基于这一框架,利用偏导数的反向传播计算方法,通 过梯度上升的优化过程来实现互信息的最大化,从而对HMM进行有区分力的训练.这一 训练方法被称之为HMM的反向传播训练方法.此外,还设计了一个用以实现这一训练方 法的在数值计算上具有强鲁棒性的算法.语音识别的实验结果证实了这一训练方法的优越 性.     

Reformulating the HMM as a trajectory model by imposing explicit relationships between static and dynamic feature vector sequences

In the present paper, a trajectory model, derived from a hidden Markov model (HMM) by imposing explicit relationships between static and dynamic feature vector sequences, is developed and evaluated. The derived model, named a trajectory HMM, can alleviate two limitations of the standard HMM, which are (i) piece-wise constant statistics within a state and (ii) conditional independence assumption of state output probabilities, without increasing the number of model parameters. In the present paper, a Viterbi-type training algorithm based on the maximum likelihood criterion is also derived. The performance of the trajectory HMM was evaluated both in speech recognition and synthesis. In a speaker-dependent continuous speech recognition experiment, the trajectory HMM achieved an error reduction over the corresponding standard HMM. Subjective listening test results showed that the introduction of the trajectory HMM improved the naturalness of synthetic speech.     

Robust speech recognition method based on discriminative environment feature extraction

It is an effective approach to learn the influence of environmental parameters,such as additive noise and channel distortions,from training data for robust speech recognition.Most of the previous methods are based on maximum likelihood estimation criterion.However,these methods do not lead to a minimum error rate result.In this paper,a novel discriinative learning method of environmental parameters,which is based on Minimum Classification Error (MCE) criterion,is proposed.In the method,a simple classifier and the Generalized Probabilistic Descent (GPD)algorithm are adopted to iteratively learn the environmental parameters.Consequently,the clean speech features are estimated from the noisy speech features with the estimated environmental parameters,and then the estimations of clean speech features are utilized in the back-end HMM classifier,Experiments show that the best error rate reudction of 32.1% is obtained,tested on a task of 18 isolated confusion Korean words,relative to a conventional HMM system.     

基于HMM和遗传神经网络的语音识别系统

本文提出了一种基于隐马尔可夫(HMM)和遗传算法优化的反向传播网络(GA-BP)的混合模型语音识别方法。该方法首先利用HMM对语音信号进行时序建模,并计算出语音对HMM的输出概率的评分,将得到的概率评分作为优化后反向传播网络的输入,得到分类识别信息,最后根据混合模型的识别算法作出识别决策。通过Matlab软件对已有的样本数据进行训练和测试。仿真结果表明,由于设计充分利用了HMM时间建模能力强和GA-BP神经网络分类能力强等特点,该混合模型比单纯的HMM具有更强的抗噪性,克服了神经网络的局部最优问题,大大提高了识别的速度,明显改善了语音识别系统的性能。     

VQ／HMM二级音节识别的研究

ＨＭＭ技术在语音识别是得到较为成功的应用，然而ＶＱ／ＨＭＭ对在词表的识别速度及识别率仍不理想，文中根据系统实现中的实时性和识别率的要求，提出了初始码本均匀法，对参加训练的各音先对其求平均，然后用各音的平均值组成初始矢量的空间，并采用码本快速迭代法以及标号直方图法与ＨＭＭ识别相结合的二级识别方法，提高了系统的识别率和识别速度，此方法简单，易于实时化。     

基于一种改进禁忌搜索算法优化离散隐马尔可夫模型

隐马尔可夫模型(HMM,HiddenMarkovModel)是语音识别和手势识别中广泛使用的统计模式识别方法。文章提出了一种改进的禁忌搜索(ITS,ImprovedTabuSearch)优化HMM的参数。传统的TabuSearch(TS)与局部搜索算法(极大似然法)交替进行,从而加快了算法的收敛速度,并得到优化解。分别用TS及ITS训练隐马尔可夫模型进行动态手势识别。结果表明ITS可获得更高的识别率,且能达到全局优化。     

重庆方言语音识别系统的设计与实现

语音识别赋予了计算机能够识别出语音内容的功能,是人机交互技术领域的重要研究内容。随着计算机技术的发展,语音识别已经得到了成熟的发展。但是关于方言的语音识别还有很大的发展空间。中国是一个幅员辽阔、人口众多的国家,因此方言种类繁多,其中有3000多万人交流使用的重庆方言就是其中之一。采集了重庆方言的部分词语的文本文件和对应的语音文件建立语料库,根据重庆方言的发音特点,选取重庆方言的声韵母作为声学建模基元,选取隐马尔可夫模型(Hidden Markov Model, HMM)为声学模型设计了一个基于HMM的重庆方言语音识别系统。在训练过程利用语料库中训练集语料对声学模型进行训练,形成HMM模型库;在识别过程利用语料库中的测试集语料进行识别测试。实验结果表明,该系统能够实现重庆方言的语音识别,并且识别的正确率为100%。