噪声环境中的汉语浊语音检测

为了在低信噪比和复杂噪声环境下检测汉语浊语音,根据浊语音谐波结构特性,提出了一种鲁棒的浊语音检测方法。通过改进的谱跟踪算法,得到能表征浊语音谐波特性的一簇谱线;从谱线簇中提取谐波特征作为汉语浊语音检测的依据。在不同信噪比和不同噪声环境下的浊语音检测对比实验中全面优于传统方法,在0 dB信噪比时正识率高于传统方法约30%。实验结果表明,该方法在低信噪比和非平稳复杂噪声环境下都具有较好的浊语音检测效果。      

改进谐波组织规则的单通道浊语音分离系统

针对以往单通道噪声和浊语音分离算法的不足,改进了谐波组织算法。算法利用载波包络能量比将时频单元分为确定和非确定。提取基频作为组织线索。组织阶段分别使用谐波原理和最小幅度原理对确定时频单元组织;使用改进包络自相关函数度量幅度调制率对非确定时频单元组织。对比以往算法的处理结果,改进算法平均信噪比(SNR)提高0.96 dB。通过对谐波组织规则的改进,提高了分离性能。     

低信噪比下采用感知语谱结构边界参数的语音端点检测算法

提出了一种采用感知语谱结构边界参数(PSSB)的语音端点检测算法,用于在低信噪比环境下的语音信号预处理。在对含噪语音进行基于听觉感知特性的语音增强之后,针对语音信号的连续分布特性与残留噪声的随机分布特性之间的不同点,对增强后语音的时-频语谱进行二维增强,从而进一步突出连续分布的纯净语音的语谱结构。通过对增强后语音语谱结构的二维边界检测,提出PSSB参数,并用于端点检测。实验结果表明,在白噪声-10 dB到10 dB的各种信噪比环境下,采用PSSB参数的端点检测算法,相对于其它端点检测算法,更有效地检测出语音的端点。在-10 dB的极低信噪比下,提出的方法仍然有75.2%的正确率。采用PSSB参数的端点检测算法,更适合于低信噪比白噪声环境下的语音端点检测。      

时频字典学习的单通道语音增强算法

针对以往语音增强算法在非平稳噪声环境下性能急剧下降的问题,基于时频字典学习方法提出了一种新的单通道语音增强算法。首先,提出采用时频字典学习方法对噪声的频谱结构的先验信息进行建模,并将其融入到卷积非负矩阵分解的框架下;然后,在固定噪声时频字典情况下,推导了时变增益和语音时频字典的乘性迭代求解公式;最后,利用该迭代公式更新语音和噪声的时变增益系数以及语音的时频字典,通过语音时频字典和时变增益的卷积运算重构出语音的幅度谱并用二值时频掩蔽方法消除噪声干扰。实验结果表明,在多项语音质量评价指标上,本文算法都取得了更好的结果。在非平稳噪声和低信噪比环境下,相比于多带谱减法和非负稀疏编码去噪算法,本文算法更有效地消除了噪声,增强后的语音具有更好的质量。      

一种基于奇异谱的语音激活检测方法

为了提高语音激活检测在低信噪比环境中的检测性能,提出了一种基于奇异谱的语音激活检测方法。首先用多窗口方法计算每一帧语音信号的相关矩阵;然后对相关矩阵进行奇异值分解;利用奇异值可以反映有用信号和噪声分布情况的特性,将每一帧语音信号经过加权处理后的最大奇异值与自适应阈值进行比较进行语音激活检测。该方法原理简单,易于硬件实现,通过实验仿真表明,在低信噪比环境下,和基于对数能量方法相比,本文方法也能够很好的区分语音段和非语音段,有良好的检测性能。     

复杂噪声环境中的语音端点检测

提出了一种适用于复杂加性噪声环境的语音端点检测方法。通过对噪声类型的归纳,建立自适应的平稳噪声模型,并根据模型搜索信号能量非平稳的区域。然后基于浊音在频域上的谐波结构,在此区域内检测浊音,从而排除非平稳噪声的干扰。最后根据信号能量精确搜索语音起止点。与目前典型端点检测算法的对比实验表明,在大多数复杂噪声环境下,该算法具有较好的准确率。     

基于谐波和能量特征的单声道浊语音分离方法

提出了一种基于谐波和能量特征的单声道浊语音分离方法。该方法将浊语音分离问题转化为声音在时频域的分类问题。首先,在已有谐波特征的基础上,引入能量特征。然后,对于谐波特征明显且能量大的时频单元,在分类器训练阶段复制它们的特征。实验结果表明该方法相比之前的方法有更好的信噪比增益。通过引入能量特征和特征复制,改善了浊语音的分离效果。      

基于线性调频Z变换和短波语音通话的飞机类型识别研究

研究用短波语音通话携带的飞机舱室噪声对飞机类型进行识别的方法。分析了飞机舱室内噪声在短波信道和语音通话干扰下的物理特性,定义了估计语音段的飞机噪声信噪比的公式,提出了自适应的抑制语音增强飞机噪声的模型,通过CZT变换分别提取目标信号不同频段的功率谱密度级特征,并设计了用支持向量机进行分类识别的二叉分类树。对8类现场实测数据进行实验:增强后语音段的平均信噪比提高约22 dB,分类树对语音应答间隔噪声、语音段信号和增强后的信号的平均识别率分别为82.79%,15.25%,50.18%。实验表明:应答间隔噪声可用于飞机类型识别;语音抑制算法带来较大的信噪比和识别率增益,证明语音段蕴含有助于飞机类型识别的重要信息,可为后续的研究奠定基础。      

滑动窗累积量的递推估计算法及其在语音端点检测中的应用

提出了一种滑动窗累积量的递推估计算法并应用于语音端点检测中,用以解决传统端点检测方法在噪声环境下检测性能变差的问题。在对含噪语音信号进行加窗之后,利用滑动窗累积量的递推估计算法估计含噪语音信号的高阶累积量值,并在此基础上结合能量特征进行语音端点检测。实验结果表明,所提滑动窗累积量递推估计算法相比较传统高阶累积量计算方法运算效率明显提高;所提端点检测算法在不同噪声和信噪比环境下相比较G.729b算法点正确率Pc-point值平均提升了6.07%。基于滑动窗高阶累积量的语音端点检测算法具有较高的运算效率及良好的鲁棒性。      

滑动窗累积量的递推估计算法及其在语音端点检测中的应用

提出了一种滑动窗累积量的递推估计算法并应用于语音端点检测中,用以解决传统端点检测方法在噪声环境下检测性能变差的问题。在对含噪语音信号进行加窗之后,利用滑动窗累积量的递推估计算法估计含噪语音信号的高阶累积量值,并在此基础上结合能量特征进行语音端点检测。实验结果表明,所提滑动窗累积量递推估计算法相比较传统高阶累积量计算方法运算效率明显提高;所提端点检测算法在不同噪声和信噪比环境下相比较G.729b算法点正确率Pc-point值平均提升了6.07%。基于滑动窗高阶累积量的语音端点检测算法具有较高的运算效率及良好的鲁棒性。     

长时语音特征在说话人识别技术上的应用

本文除介绍常用的说话人识别技术外,主要论述了一种基于长时时频特征的说话人识别方法,对输入的语音首先进行VAD处理,得到干净的语音后,对其提取基本时频特征。在每一语音单元内把基频、共振峰、谐波等时频特征的轨迹用Legendre多项式拟合的方法提取出主要的拟合参数,再利用HLDA的技术进行特征降维,用高斯混合模型的均值超向量表示每句话音时频特征的统计信息。在NIST06说话人1side-1side说话人测试集中,取得了18.7%的等错率,与传统的基于MFCC特征的说话人系统进行融合,等错率从4.9%下降到了4.6%,获得了6%的相对等错率下降。      

A recursive calculating algorithm for higher-order cumulants over sliding window and its application in speech endpoint detection

Regarding the performance of traditional endpoint detection algorithms degrades as the environment noise level increases,a recursive calculating algorithm for higher-order cumulants over a sliding window is proposed.Then it is applied to the speech endpoint detection.Furthermore,endpoint detection is carried out with the feature of energy.Experimental results show that both the computational efficiency and the robustness against noise of the proposed algorithm are improved remarkably compared with traditional algorithm.The average probability of correct point detection(Pc-point) of the proposed voice activity detection(VAD) is6.07%higher than that of G.729 b VAD in different noisy at different signal-noise ratios(SNRs)environments.     

Speech endpoint detection in real noise environments

A method of speech endpoint detection in environments of complicated additive noise is presented. Based on the analysis of noise, an adaptive model of stationary noise is proposed to detect the section where the signal is nonstationary. Then the voice is detected in this section by its harmonic structure, and the accurate endpoint is searched using energy. Compared with the typical algorithms, this algorithm operates reliably in most real noise environments.     

An effective cluster-based model for robust speech detection and speech recognition in noisy environments

This paper shows an accurate speech detection algorithm for improving the performance of speech recognition systems working in noisy environments. The proposed method is based on a hard decision clustering approach where a set of prototypes is used to characterize the noisy channel. Detecting the presence of speech is enabled by a decision rule formulated in terms of an averaged distance between the observation vector and a cluster-based noise model. The algorithm benefits from using contextual information, a strategy that considers not only a single speech frame but also a neighborhood of data in order to smooth the decision function and improve speech detection robustness. The proposed scheme exhibits reduced computational cost making it adequate for real time applications, i.e., automated speech recognition systems. An exhaustive analysis is conducted on the AURORA 2 and AURORA 3 databases in order to assess the performance of the algorithm and to compare it to existing standard voice activity detection (VAD) methods. The results show significant improvements in detection accuracy and speech recognition rate over standard VADs such as ITU-T G.729, ETSI GSM AMR, and ETSI AFE for distributed speech recognition and a representative set of recently reported VAD algorithms.     

A subspace approach based on embedded prewhitening for voice activity detection

This paper presents a subspace approach for voice activity detection (VAD). The proposed approach is based on an embedded prewhitening scheme for the simultaneous diagonalization of the clean speech and noise covariance matrices to provide a decision rule based on likelihood ratio test in signal subspace domain. Experimental results show that the proposed subspace-based VAD algorithm outperforms the method using a Gaussian model in a conventional discrete Fourier transform domain at the low signal-to-noise conditions.     

Optimally weighted maximum a posteriori probabilities based on minimum classification error for dual-microphone voice activity detection

The dual-microphone voice activity detection (VAD) technique is proposed by applying discriminative weight training to achieve optimal weighting of spatial features available within the dual-microphone VAD. Since the motivation behind our method is to use the relevant spatial information available from the two microphones, we employ the phase difference, coherence, and power level difference ratio (PLDR) as a feature vector, and then use this feature vector to derive the maximum a posteriori (MAP) probabilities. Then, we combine each MAP probability based on a discriminative weight training, i.e., the minimum classification error (MCE) method to offer an optimal VAD decision in a spectral domain, which successfully represents the dynamic evolution of speech over time even in the non-stationary noise environments. The proposed dual-microphone VAD algorithm outperforms conventional dual-microphone VAD methods based on only single feature among the PLDR, phase difference, and spectral coherence.     

仿选择性注意机制的语音情感识别算法

有效特征的选取一直都是语音情感识别算法的关键。为此,针对语音情感特征选择与构建的问题,一种仿选择性注意机制的语音情感识别算法被提出。考虑到语音信号的时频特性,算法首先计算语音信号的语谱图;其次,模仿选择性注意机制,计算语谱图的颜色、方向和亮度特征图,归一化后形成特征矩阵;然后,将特征矩阵重排列并进行PCA降维,形成情感识别特征向量;最后,利用改进的支持向量机分类方法进行语音情感识别。对愤怒、恐惧、高兴、悲伤和惊奇5种情感的识别实验显示,基于选择性注意的方法能够获得较好的识别效果,平均识别率为85.44%。相比于韵律特征和音质特征,语音情感识别率至少提高10%;相比于其它语谱特征,识别率提高7%左右。