利用多层感知机映射提高不匹配环境下的语音识别性能

本文提出了一种提高不匹配环境下的语音识别性能方法。实验证明,采用了本文提出的算法后,测试环境下的语音识别性能有了很大的提高。     

自适应并行模型组合的鲁棒语音身份识别算法

由于环境噪声的影响,实际应用中说话人识别系统性能会出现急剧下降。提出了一种基于高斯混合模型-通用背景模型和自适应并行模型组合的鲁棒性语音身份识别方法。自适应并行模型组合是一种噪声鲁棒性的特征补偿算法,能够有效减少训练环境与测试环境之间的不匹配现象,从而提高系统识别准确率和抗噪性能。首先,算法从测试语音中估计出噪声特征,然后用一个单高斯模型对噪声特征进行拟合得到噪声均值和协方差。最后,根据得出的噪声均值和协方差,调整训练好的高斯混合模型均值向量和协方差矩阵,使其尽可能地匹配测试环境。实验结果表明,该方法可以准确地重构干净语音的高斯混合模型参数,并且能够显著提高说话人识别的准确率,特别是在低信噪比情况下。      

基于矢量泰勒级数的模型自适应算法

在实际环境中,由于测试环境与训练环境的不匹配,语音识别系统的性能会急剧恶化。模型自适应算法是减小环境失配影响的有效方法之一,它通过测试环境下的少量自适应数据,将HMM模型的参数变换到测试环境下。该文将矢量泰勒级数用于模型自适应,同时对HMM模型的均值向量和协方差矩阵进行变换,使其与实际环境相匹配。实验证明,该文算法优于MLLR算法和基于矢量泰勒级数的特征补偿算法,在低信噪比环境中性能提高尤为明显。     

鲁棒性语音识别技术研究综述

鲁棒性语音识别是为了解决噪声环境和混响环境等外界因素所引起的语音识别系统训练和识别不匹配的情况,针对在噪声和混响条件下进行鲁棒性语音识别的问题,对现有的鲁棒性语音识别研究进行了总结,阐述了语音识别的主要流程和整体框架;从特征提取和声学建模两个方面对语音增强技术、语音分离技术进行介绍;分析鲁棒性语音识别技术主要难点和实现过程。最后对鲁棒性语音识别技术进行总结和展望。     

语音增强用于抗噪声的汉语说话人识别

大多数实际应用环境中总是存在各种各样的噪声，由于训练环境与识别环境不匹配，现有的绝大多数说话人识别系统在噪声环境中的性能都不可避免的急剧下降。为了让说话人识别系统在强噪声环境中，有较好的识别效果．研究一个将语音增强器和说话人识别系统级连起来的系统，该系统中将语音增强作为前端处理来提高输入的信噪比。实验证明，该系统具有很好的抗噪声性能。     

在线更新噪声基矩阵的非负矩阵分解语音增强方法

基于非负矩阵分解(Nonnegative matrix factorization, NMF)的语音增强算法需要和背景噪声类型匹配的噪声基矩阵(Basis matrix)，而在实际中，这是很难被保证的。本文提出了一种基于噪声基矩阵在线更新的非负矩阵分解语音增强方法，该方法首先利用一个无语音帧判决模块识别出带噪语音的无语音区域，然后利用一个固定长度的滑动窗口(Sliding window)来包含若干帧最近过去的带噪语音的无语音帧，并用这些无语音帧的幅度谱在线更新噪声基矩阵，最后利用更新得到的噪声基矩阵和预先训练的语音基矩阵实现语音增强。该方法能够在线更新出匹配的噪声基矩阵，有效地解决了噪声基矩阵不匹配的问题。实验证明，本文所提的方法在线学习到的噪声基矩阵在大多数条件下比匹配训练集下训练得到的噪声基矩阵的性能还要优越。      

语音识别中环境失配补偿综述

作为语音处理领域的支撑技术之一,语音识别以识别语音信号并将其转变成文字为目标,在智能人机交互、对话系统、多媒体内容分析等领域有着广阔的应用前景。经过数十年的发展,目前的语音识别技术在理想状况下能取得较高的识别率。然而,在采集和传输过程中,语音信号不可避免地会受到各种信道和加性噪声的干扰,引起训练环境和识别环境不一致、即环境失配,进而导致识别系统的性能急剧下降。这种失配严重阻碍了语音识别技术走向现实应用,已成为语音识别领域中迫切需要解决的问题之一。首先阐述了环境失配的问题,然后按照加性噪声、信道畸变和联合补偿的脉络,系统地综述了各个问题的补偿方法。     

语音识别的互信息匹配模型及其应用

互信息是随机变量或随机信号之间所携带对方信息的定量描述。语音识别的互信息匹配模型依据互信息理论以及语音信号特征参数统计分析结果而提出,这一模型同时考虑了语音信号的时变发布特征和统计分布特征,能有效地提高同类模式的凝聚度,减少非同类模式间的耦合性。语音识别实验和实际应用表明,互信息匹配模型具有良好的识别精度和很高的运行效率,与其它模式匹配方法相比更适合嵌入式系统的语音识别应用。对互信息匹配模型的原理和特点进行了全面的阐述,并介绍了这一模型在几个语音识别系统中的具体应用。     

听觉注意模型的语谱图语音情感识别方法

在语音情感识别技术中,由于噪声环境、说话方式和说话人特质原因,会造成实验数据库特征不匹配的情况。从语音学上分析,该问题多存在于跨数据库情感识别实验。训练的声学模型和用于测试的语句样本之间的错位,会使语音情感识别性能剧烈下降。本文据此所研究的选择性注意声学模型能有效探测变化的情感特征。同时,利用时频原子对模型进行改进,使之能提取跨语音数据库中的显著性特征用于情感识别。实验结果表明,利用文章所提方法在跨库情感样本上进行特征提取,再通过典型的分类器,识别性能提高了9个百分点,从而验证了该方法对不同数据库具有更好的鲁棒性。      

具有环境自学习机制的鲁棒说话人识别算法

说话人识别系统实际应用时,一旦应用环境和训练环境不一致,系统的性能会急剧下降。由于环境噪声的多变性,系统训练时无法预测实际应用中的环境噪声。因此,引入环境自学习和自适应思想,通过改进的矢量泰勒级数(Vector Taylor Series,VTS)刻画环境噪声模型和说话人语音模型之间的统计关系,提出一种具有环境自学习能力的鲁棒说话人识别算法。系统应用中每当环境变化时利用语音输入前采集到的环境噪声信号来迭代更新环境噪声模型参数,进一步基于VTS确立的统计关系,将说话人语音模型自适应到实际应用环境来补偿环境失配的影响。说话人辨认实验结果表明,提出的方法在低信噪比条件下对于不同种类的噪声都能显著提升系统的识别性能。     

基于加权特征值补偿的说话人识别

背景噪声的存在,使得说话人识别系统的训练环境和测试环境发生失配,导致系统性能发生急剧下降。本论文提出一种加权特征值补偿算法,把由噪声引起的使带噪语音信号特征值与纯净语音特征值发生偏差的部分去除,从而使进入识别器的特征值接近纯净语音的特征值。在特征值补偿过程中引入了信噪比加权的方法。实验表明,这种方法能够有效的提高说话人识别系统的性能。     

基于VEMAP的说话人识别鲁棒性研究

为了改善发声力度变化对说话人识别系统性能的影响.针对不同发声力度下语音信号的分析,提出了使用发声力度最大后验概率(Vocal Effort Maximum A Posteriori,VEMAP)自适应方法更新基于高斯混合模型-通用背景模型(Gaussian Mixture Model-Universal Background Model,GMM-UBM)的说话人识别系统模型.实验表明,所提出的方法使不同发声力度下系统EER％降低了88.45％与85.16％,有效解决了因发声力度变化引起的训练语音与测试语音音量失配,从而导致说话人识别性能降低的问题,改善说话人识别系统性能效果显著.     

Improved perceptually non-uniform spectral compression for robust speech recognition

According to the decline of recognition rate of speech recognition system in the noise environments, an improved perceptually non-uniform spectral compression feature extraction algorithm is put forward in this paper. This method can realize an effective compression of the speech signals and make the training and recognition environments more matching, so the recognition rate can be improved in the noise environments. By experimenting on the intelligent wheelchair platform, the result shows that the algorithm can effectively enhance the robustness of speech recognition, and ensure the recognition rate in the noise environments.     

On adaptive decision rules and decision parameter adaptation forautomatic speech recognition

Recent advances in automatic speech recognition are accomplished by designing a plug-in maximum a posteriori decision rule such that the forms of the acoustic and language model distributions are specified and the parameters of the assumed distributions are estimated from a collection of speech and language training corpora. Maximum-likelihood point estimation is by far the most prevailing training method. However, due to the problems of unknown speech distributions, sparse training data, high spectral and temporal variabilities in speech, and possible mismatch between training and testing conditions, a dynamic training strategy is needed. To cope with the changing speakers and speaking conditions in real operational conditions for high-performance speech recognition, such paradigms incorporate a small amount of speaker and environment specific adaptation data into the training process. Bayesian adaptive learning is an optimal way to combine prior knowledge in an existing collection of general models with a new set of condition-specific adaptation data. In this paper, the mathematical framework for Bayesian adaptation of acoustic and language model parameters is first described. Maximum a posteriori point estimation is then developed for hidden Markov models and a number of useful parameters densities commonly used in automatic speech recognition and natural language processing     

并行子带HMM最大后验概率自适应非线性类估计算法

目前,自动语音识别(ASR)系统在实验室环境下获得了较高的识别率,但是在实际环境中,由于受到背景噪声和传输信道的影响,系统的识别性能急剧恶化.本文以听觉试验为基础,提出一种新的独立子带并行最大后验概率的非线性类估计算法,用以提高识别系统的鲁棒性.本算法利用多种噪声和识别内容功率谱差异,以及噪声在不同频带上对HMM影响的不同,采用多层感知机(MLP)对噪声环境下最大后验概率进行非线性映射,以减少识别系统由于环境不匹配而导致的识别性能下降.实验表明:该算法性能明显优于最大后验线性回归算法和Sangita提出的子带语音识别算法.     

Human emotion recognition by optimally fusing facial expression and speech feature

Emotion recognition is a hot research in modern intelligent systems. The technique is pervasively used in autonomous vehicles, remote medical service, and human–computer interaction (HCI). Traditional speech emotion recognition algorithms cannot be effectively generalized since both training and testing data are from the same domain, which have the same data distribution. In practice, however, speech data is acquired from different devices and recording environments. Thus, the data may differ significantly in terms of language, emotional types and tags. To solve such problem, in this work, we propose a bimodal fusion algorithm to realize speech emotion recognition, where both facial expression and speech information are optimally fused. We first combine the CNN and RNN to achieve facial emotion recognition. Subsequently, we leverage the MFCC to convert speech signal to images. Therefore, we can leverage the LSTM and CNN to recognize speech emotion. Finally, we utilize the weighted decision fusion method to fuse facial expression and speech signal to achieve speech emotion recognition. Comprehensive experimental results have demonstrated that, compared with the uni-modal emotion recognition, bimodal features-based emotion recognition achieves a better performance.     

Class‐Based Histogram Equalization for Robust Speech Recognition

A new class‐based histogram equalization method is proposed for robust speech recognition. The proposed method aims at not only compensating the acoustic mismatch between training and test environments, but also at reducing the discrepancy between the phonetic distributions of training and test speech data. The algorithm utilizes multiple class‐specific reference and test cumulative distribution functions, classifies the noisy test features into their corresponding classes, and equalizes the features by using their corresponding class‐specific reference and test distributions. Experiments on the Aurora 2 database proved the effectiveness of the proposed method by reducing relative errors by 18.74%, 17.52%, and 23.45% over the conventional histogram equalization method and by 59.43%, 66.00%, and 50.50% over mel‐cepstral‐based features for test sets A, B, and C, respectively.