A clustering based feature selection method in spectro-temporal domain for speech recognition

Spectro-temporal representation of speech has become one of the leading signal representation approaches in speech recognition systems in recent years. This representation suffers from high dimensionality of the features space which makes this domain unsuitable for practical speech recognition systems. In this paper, a new clustering based method is proposed for secondary feature selection/extraction in the spectro-temporal domain. In the proposed representation, Gaussian mixture models (GMM) and weighted K-means (WKM) clustering techniques are applied to spectro-temporal domain to reduce the dimensions of the features space. The elements of centroid vectors and covariance matrices of clusters are considered as attributes of the secondary feature vector of each frame. To evaluate the efficiency of the proposed approach, the tests were conducted for new feature vectors on classification of phonemes in main categories of phonemes in TIMIT database. It was shown that by employing the proposed secondary feature vector, a significant improvement was revealed in classification rate of different sets of phonemes comparing with MFCC features. The average achieved improvements in classification rates of voiced plosives comparing to MFCC features is 5.9% using WKM clustering and 6.4% using GMM clustering. The greatest improvement is about 7.4% which is obtained by using WKM clustering in classification of front vowels comparing to MFCC features.     

Effect of aging on speech features and phoneme recognition: a study on Bengali voicing vowels

The article studies age related variations of speech characteristics of two age groups, in the Bengali language. The study considers 60 speakers in the each age groups, 60–80 years and 20–40 years, respectively. We have considered different voice source features like fundamental frequency, formant frequencies, jitter, shimmer and harmonic to noise ratio. Cepstral domain feature, Mel Frequency Cepstral coefficients (MFCC) of different voiced Bengali vowels are also analyzed for younger and older adult groups. MFCC feature and Hidden Markov model parameter of different voiced vowels are used to study phoneme dissimilarities measure between two age groups. Age related changes in elderly speech affect the automatic speech recognition performance as was observed in our study, raising the need for specific acoustic models for elderly persons.     

SMFCC:一种新的语音信号特征提取方法

针对说话人识别系统中存在的有效语音特征提取以及噪声影响的问题,提出了一种新的语音特征提取方法——基于S变换的美尔倒谱系数(SMFCC)。该方法是在传统美尔倒谱系数(MFCC)的基础上利用S变换的二维时频多分辨率特性,以及奇异值分解(SVD)方法的二维时频矩阵有效去噪性,并结合相关统计分析方法最终获得语音特征。采用TIMIT语音数据库,将所提的特征和现有特征进行对比实验。SMFCC特征的等错误率(EER)和最小检测代价(MinDCF)均小于线性预测倒谱系数(LPCC)、MFCC及其结合方法LMFCC,比MFCC的EER和MinDCF08分别下降了3.6%与17.9%。实验结果表明所提方法能够有效去除语音信号中的噪声,提升局部分辨率。     

Self-Adjustable Neural Network for speech recognition

Neural networks with fixed input length are not able to train and test data with variable lengths in one network size. This issue is very crucial when the neural networks need to deal with signals of variable lengths, such as speech. Though various methods have been proposed in segmentation and feature extraction to deal with variable lengths of the data, the size of the input data to the neural networks still has to be fixed. A novel Self-Adjustable Neural Network (SANN) is presented in this paper, to enable the network to adjust itself according to different data input sizes. The proposed method is applied to the speech recognition of Malay vowels and TIMIT isolated words. SANN is benchmarked against the standard and state-of-the-art recogniser, Hidden Markov Model (HMM). The results showed that SANN was better than HMM in recognizing the Malay vowels. However, HMM outperformed SANN in recognising the TIMIT isolated words.     

基于听皮层神经元感受野的强噪声环境下说话人识别

针对说话人识别易受环境噪声影响的问题,借鉴生物听皮层神经元频谱-时间感受野（STRF）的时空滤波机制,提出一种新的声纹特征提取方法。在该方法中,对基于STRF获得的听觉尺度-速率图进行了二次特征提取,并与传统梅尔倒谱系数（MFCC）进行组合,获得了对环境噪声具有强容忍的声纹特征。采用支持向量机（SVM）作为分类器,对不同信噪比（SNR）语音数据进行测试的结果表明,基于STRF的特征对噪声的鲁棒性普遍高于MFCC系数,但识别正确率较低;组合特征提升了语音识别的正确率,同时对环境噪声具有良好的鲁棒性。该结果说明所提方法在强噪声环境下说话人识别上是有效的。     

基于听皮层神经元感受野的强噪声环境下说话人识别

针对说话人识别易受环境噪声影响的问题，借鉴生物听皮层神经元频谱-时间感受野（STRF）的时空滤波机制，提出一种新的声纹特征提取方法。在该方法中，对基于STRF获得的听觉尺度-速率图进行了二次特征提取，并与传统梅尔倒谱系数（MFCC）进行组合，获得了对环境噪声具有强容忍的声纹特征。采用支持向量机（SVM）作为分类器，对不同信噪比（SNR）语音数据进行测试的结果表明，基于STRF的特征对噪声的鲁棒性普遍高于MFCC系数，但识别正确率较低；组合特征提升了语音识别的正确率，同时对环境噪声具有良好的鲁棒性。该结果说明所提方法在强噪声环境下说话人识别上是有效的。     

基于特征优选和字典优化的组稀疏表示表情识别

针对在小样本人脸表情数据库上识别模型过拟合问题,文中提出基于特征优选和字典优化的组稀疏表示分类方法.首先提出特征优选准则,选择相同类级稀疏模式、不同类内稀疏模式的互补特征构建字典.然后对字典进行最大散度差优化学习,使字典在不失真重构特征的同时具有较高鉴别能力.最后联合优化后的字典进行组稀疏表示分类.在JAFFE、CK+数据库上的实验表明,文中方法对样本减少具有鲁棒性,泛化能力较强,识别精度较优.     

Filterbank optimization for robust ASR using GA and PSO

Automatic speech recognition (ASR) systems follow a well established approach of pattern recognition, that is signal processing based feature extraction at front-end and likelihood evaluation of feature vectors at back-end. Mel-frequency cepstral coefficients (MFCCs) are the features widely used in state-of-the-art ASR systems, which are derived by logarithmic spectral energies of the speech signal using Mel-scale filterbank. In filterbank analysis of MFCC there is no consensus for the spacing and number of filters used in various noise conditions and applications. In this paper, we propose a novel approach to use particle swarm optimization (PSO) and genetic algorithm (GA) to optimize the parameters of MFCC filterbank such as the central and side frequencies. The experimental results show that the new front-end outperforms the conventional MFCC technique. All the investigations are conducted using two separate classifiers, HMM and MLP, for Hindi vowels recognition in typical field condition as well as in noisy environment.     

融合倒谱特征的脑电（EEG）情感分类

近年来,通过分析脑电图（EEG）信号来实现情感识别的课题越来越被研究者所重视。为了丰富特征的表示能力,获得更高的情感识别分类准确率,尝试将语音信号特征梅尔频率倒谱系数MFCC应用于脑电信号。在对EEG信号小波变换的基础上将提取得到的MFCC特征与EEG特征相互融合,通过利用深度残差网络（ResNet18）的特性进行情感分类识别。实验结果表明,比起传统的单一利用EEG特征,添加了MFCC特征使得情感维度Arousal和Valence两者的识别准确率分别提升了6%和4%,达到了86.01%和85.46%,从而提升了情感的识别准确度。     

基于元音检测的汉语连续语音端点检测方法

在汉语连续语音识别中,准确检测出音节的始点和终点是很重要的一步,传统的端点检测方法在非连续语音中检测准确度很高,但在连续语音中检测准确度会大幅度降低。利用MFCC0参数和汉语元音的共振峰能量设计了一种新的端点检测方法,可以准确检测出汉语连续语音中的音节端点。实验结果表明：这种端点检测方法在低信噪比下也有很高的检测正确率。     

采用GW-MFCC模型空间参数的语音情感识别

针对单一语音特征对语音情感表达不完整的问题,将具有良好量化和插值特性的LSF参数与体现人耳听觉特性的MFCC参数相融合,提出基于线谱权重的MFCC（WMFCC）新特征。同时,通过高斯混合模型来对该参数建立模型空间,进一步得到GW-MFCC模型空间参数,以获取更高维的细节信息,进一步提高情感识别性能。采用柏林情感语料库进行验证,新参数的识别率比传统的MFCC和LSF分别有5.7%和6.9%的提高。实验结果表明,提出的WMFCC以及GW-MFCC参数可以有效地表现语音情感信息,提高语音情感识别率。     

全局和时序结构特征并用的语音信号情感特征识别方法

在利用全局特征进行语音情感特征分析的基础上,提出了采用情感语句中各元音时序 结构作为新的特征,并针对不同语句中包含不同元音个数的情况,提出了零补齐、分局均值补 齐、前均值补齐三种不同的规整方法.以从10名话者中搜集的带有欢快、愤怒、惊奇、悲伤4种 情感的1000句语句作为样本,本文对语音情感特征进行了分析.实验结果表明利用全局特征和 时序特征相结合,对时序特征采用前均值补齐,同时使用修正二次判别函数(MQDF)进行情感 识别能够获得94%的平均情感识别率.     

基于Fisher比的梅尔倒谱系数混合特征提取方法

针对语音识别中梅尔倒谱系数（MFCC）对中高频信号的识别精度不高,并且没有考虑各维特征参数对识别结果影响的问题,提出基于MFCC、逆梅尔倒谱系数（IMFCC）和中频梅尔倒谱系数（MidMFCC）,并结合Fisher准则的特征提取方法。首先对语音信号提取MFCC、IMFCC和MidMFCC三种特征参数,分别计算三种特征参数中各维分量的Fisher比,通过Fisher比对三种特征参数进行选择,组成一种混合特征参数,提高语音中高频信息的识别精度。实验结果表明,在相同环境下,新的特征与MFCC参数相比,识别率有一定程度的提高。     

基于Gabor滤波的语音识别鲁棒性研究

为了提高语音识别系统的鲁棒性,提出一种基于GBFB（spectro-temporal Gabor filter bank）的声学特征提取方法,并通过分块PCA算法对高维的GBFB特征进行降维处理,最后在多个相同噪音环境对GBFB特征以及常用的GFCC,MFCC,LPCC等特征进行抗噪性能对比,与GFCC相比GBFB特征的识别率提高了5.35%,与MFCC特征相比提升了7.05%,比LPCC特征识别的基线低9个分贝。实验结果表明,在噪音环境下与传统的GFCC、MFCC以及LPCC等特征相比GBFB特征有更优越的鲁棒性。     

Autoregressive modeling of speech trajectory transformed to the reconstructed phase space for ASR purposes

Investigating new effective feature extraction methods applied to the speech signal is an important approach to improve the performance of automatic speech recognition (ASR) systems. Owing to the fact that the reconstructed phase space (RPS) is a proper field for true detection of signal dynamics, in this paper we propose a new method for feature extraction from the trajectory of the speech signal in the RPS. This method is based upon modeling the speech trajectory using the multivariate autoregressive (MVAR) method. Moreover, in the following, we benefit from linear discriminant analysis (LDA) for dimension reduction. The LDA technique is utilized to simultaneously decorrelate and reduce the dimension of the final feature set. Experimental results show that the MVAR of order 6 is appropriate for modeling the trajectory of speech signals in the RPS. In this study recognition experiments are conducted with an HMM-based continuous speech recognition system and a naive Bayes isolated phoneme classifier on the Persian FARSDAT and American English TIMIT corpora to compare the proposed features to some older RPS-based and traditional spectral-based MFCC features.     

基于EMD的改进MFCC的语音情感识别

人在不同情感下的语音信号其非平稳性尤为明显,传统的MFCC只能反映语音信号的静态特征,经验模态分解能够精细地刻画语音信号的非平稳特性。为提取情感语音的非平稳特征,用经验模态分解将情感语音信号分解为一系列固有模态函数分量,通过Mel滤波器后取其对数能量,进行DCT反变换后得到改进的MFCC作为情感识别的新特征,采用支持向量机对高兴、生气、厌烦和恐惧等四种语音情感识别。仿真实验结果表明：改进的MFCC识别率达到77.17%,在不同的信噪比下,识别率最大可提高3.26%。     

基于时频分布与MFCC的说话人识别

针对MFCC不能得到高效的说话人识别性能的问题,提出了将时频特征与MFCC相结合的说话人特征提取方法。首先得到语音信号的时频分布,然后将时频域转换到频域再提取MFCC+MFCC作为特征参数,最后通过支持向量机来进行说话人识别研究。仿真实验比较了MFCC、MFCC+MFCC分别作为特征参数时语音信号与各种时频分布的识别性能,结果表明基于CWD分布的MFCC和MFCC的识别率可提高到95.7%。     

基于子带信息的鲁棒语音特征提取框架

本文提出一种鲁棒语音特征提取框架。通过使用一种基于子带能量分布的噪声估计方法,无需静音段,就可以估计出带噪语音的子带噪声,同时提出结合谱减和谱加权方法对特征进行处理,最终生成具有较高鲁棒性的特征。 实验证明,在语音识别系统中,这种特征可以有效提高语音识别的鲁棒性,在噪声较强(信噪比0dB到15dB)的情况下,识别率可以提高20%以上;并且,在干净语音的情况下又能保证识别率没有大的下降;同时,这种特征上的处理方法对各种噪声的适应能力都很强,无需对噪声进行预先分类即可得到很好的抗噪效果。