基于隐马尔可夫模型的非侵入式负荷监测泛化性能改进

隐马尔可夫模型(HMM)是非侵入式负荷监测常用的算法.由于电压波动与负荷自身电气特性变化等原因,负荷的测量状态如功率可能持续变化,运行过程中出现新的状态转移,但当前基于HMM的非侵入式负荷监测方法并未考虑如何处理该情况,缺乏状态辨识与功率分解的泛化能力.针对这一问题,本文提出并构建二元参数隐马尔科夫模型(BPHMM),结合DBSCAN聚类算法,基于有功功率和稳态电流对负荷状态进行聚类,降低了因电压波动和噪声数据对负荷状态聚类结果造成干扰的可能性;改进维特比算法使其考虑到HMM模型参数更新以实现对负荷状态预测泛化性能的改进;考虑到功率的随机波动性,基于极大似然估计原理构建功率计算优化模型并实现负荷的功率分解.本文采用公共数据集AMPds2对所述方法进行验证,测试算例验证了所述方法的有效性.     

基于麦克风阵和ICA的语音增强技术

介绍了极大似然估计独立分量分析的基本理论。在此基础上提出了一种将麦克风阵和独立分量分析模型的源分布自适应极大似然估计算法相结合的技术，并成功地用于实际环境中的麦克风阵带噪语音信号，实现了语音和噪声的分离，增强了语音。试验和仿真结果证明，这种算法是有效的。     

基于EM算法的图像小波系数统计研究

基于小波分析的贝叶斯(Bayes)图像处理方法常常需要获得图像小波波系数的先验概率分布密度,该文提出,利用混合高斯模型对正交小波域中自然图像的父子小波系数的联合分布密度进行建模,运用非完备数据的极大似然估计算法——期望极大(EM)算法,对该模型的参数进行估计并且给出了联合分布密度函数的模型分量数与迭代次数的确定过程。最后,在后验均值(PM)方法下,把该联合分布密度模型运用于图像去噪研究;仿真结果表明该方法能够获得较好的效果。     

基于增量式有限混合模型的多目标状态极大似然估计

提出了增量式有限混合模型来提取概率假设密度滤波器序贯蒙特卡罗实现方式中的多目标状态. 该模型以增量方式构建, 其混合分量采用逐个方式插入其中. 采用极大似然准则来估计多目标状态. 对于给定分量数目的混合模型, 应用期望极大化算法来获得参数的极大似然解. 在新分量插入混合模型时, 保持已有混合模型的参数不变, 仍旧采用极大似然准则从候选新分量集合中选择新插入分量. 新分量插入混合步和期望极大化算法拟合混合参数步交替应用直到混合分量数目达到概率假设密度滤波器的目标数目估计值. 利用k-d树生成插入到混合模型的新分量候选集合. 增量式有限混合模型统一了分量数目变化趋势和粒子集合似然函数的变化趋势, 有助于一步一步地搜寻混合模型的极大似然解. 仿真结果表明, 基于增量式有限混合模型的概率假设密度滤波器状态提取算法在多目标跟踪的应用中优于已有的状态提取算法.     

一种改进的最小统计噪声功率谱估计算法

噪声功率谱估计是语音增强系统中的一个重要部分。基于Martin提出的最小统计噪声功率谱估计算法（MS）提出了一种改进的噪声功率谱估计算法。实验结果表明算法能够较好跟踪噪声谱的变化,提高噪声功率谱估计的准确性,改善增强后语音的质量。     

多个三维刚体运动参数的鲁棒估计算法

本文提出了一种估计多个三维刚体运动参数的鲁棒算法，可以处理包含高斯噪声和出格点的对应点数据，根据贝叶斯统计决策规则和蕴含在问题中的启发式规则，我们将运动参数估计问题转化为极大似然估计过程，实现部分模型拟合。因此，这种优化算法就是估计一组三维运动参数，使对应点数据最大限度地拟合似然函数，从而保证算法的鲁棒性。     

语音分离与HMM相结合的语音增强方法

针对基于隐马尔科夫（HMM,Hidden Markov Model）的MAP和MMSE两种语音增强算法计算量大且前者不能处理非平稳噪声的问题,借鉴语音分离方法,提出了一种语音分离与HMM相结合的语音增强算法。该算法采用适合处理非平稳噪声的多状态多混合单元HMM,对带噪语音在语音模型和噪声模型下的混合状态进行解码,结合语音分离方法中的最大模型理论进行语音估计,避免了迭代过程和计算量特别大的公式计算,减少了计算复杂度。实验表明,该算法能够有效地去除平稳噪声和非平稳噪声,且感知评价指标PESQ 的得分有明显提高,算法时间也得到有效控制。     

EM算法在统计自然语言处理中的应用

在统计自然语言处理中会经常遇到一类参数估值问题,就是当观察数据为不完全数据时如何求解参数的最大似然估计,EM算法就是解决这类问题的经典算法.给出了EM算法的基本框架,结合HMM和PCFG模型给出如何应用EM算法求解参数的极大似然估计,讨论了EM算法的优点和不足之处.     

基于极大似然准则和最大期望算法的自适应UKF 算法

针对噪声先验统计特性未知情况下的非线性系统状态估计问题,提出了基于极大似然准则和 最大期望算法的自适应无迹卡尔曼滤波(Unscented Kalman filter, UKF) 算法.利用极大似然准则构造含有噪声统计特性的对数似然函数,通 过最大期望算法将噪声估计问题转化为对数似然函数数学期望极大化问题,最终得到带次优递 推噪声统计估计器的自适应UKF算法.仿真分析表明,与传统UKF算法相比,提出的自适应UKF算法 有效克服了传统UKF算法在系统噪声统计特性未知情况下滤波精度下降的问题,并实现了系统噪 声统计特性的在线估计.     

基于间接拉普拉斯模型因子估计的语音增强算法

对DCT城基于拉普拉斯统计模型的语音增强，分析了模型因子的估计误差及其对于算法整体增强性能的影响，并根据广义高斯分布模型度其形态参数的概念与性质，提出了一种新的拉普拉斯模型因子估计方法，该方法结构简单，它利用拉普拉斯模型条件下语音分量方差与模型因子的对应关系，间接地获取模型因子的估计，算法不仅有效地消除了噪声分量对于估计精度的影响，而且可以快速地跟踪语音分量的变化。仿真结果表明，基于该模型因子估计方法的语音增强算法在多种噪声背景下具有更出色的语音增强效果。     

Change Point Detection in GARCH Models for Voice Activity Detection

This paper presents a robust algorithm for voice activity detection (VAD) based on change point detection in a generalized autoregressive conditional heteroscedasticity (GARCH) process. GARCH models are new statistical methods that are used especially in economic time series and are a popular choice to model speech signals and their changing variances. Change point detection is also important in economic sciences. In this paper, no distinct probability functions are assumed for speech and noise distributions. Also, to detect speech/nonspeech intervals, no likelihood ratio test (LRT) is employed. For testing parameter constancy in GARCH models, the algorithm of the Cramer-von Mises (CVM) test is described. This test is a nonparametric test and is based on the empirical quantiles. We show that VAD is related to the parameter constancy test in GARCH process, and we illustrate several examples.     

Integrated Speech Enhancement Method Using Noise Suppression and Dereverberation

This paper proposes a method for enhancing speech signals contaminated by room reverberation and additive stationary noise. The following conditions are assumed. 1) Short-time spectral components of speech and noise are statistically independent Gaussian random variables. 2) A room's convolutive system is modeled as an autoregressive system in each frequency band. 3) A short-time power spectral density of speech is modeled as an all-pole spectrum, while that of noise is assumed to be time-invariant and known in advance. Under these conditions, the proposed method estimates the parameters of the convolutive system and those of the all-pole speech model based on the maximum likelihood estimation method. The estimated parameters are then used to calculate the minimum mean square error estimates of the speech spectral components. The proposed method has two significant features. 1) The parameter estimation part performs noise suppression and dereverberation alternately. (2) Noise-free reverberant speech spectrum estimates, which are transferred by the noise suppression process to the dereverberation process, are represented in the form of a probability distribution. This paper reports the experimental results of 1500 trials conducted using 500 different utterances. The reverberation time RT₆₀ was 0.6 s, and the reverberant signal to noise ratio was 20, 15, or 10 dB. The experimental results show the superiority of the proposed method over the sequential performance of the noise suppression and dereverberation processes.     

Quasi-Monte Carlo estimation in generalized linear mixed models

Generalized linear mixed models (GLMMs) are useful for modelling longitudinal and clustered data, but parameter estimation is very challenging because the likelihood may involve high-dimensional integrals that are analytically intractable. Gauss–Hermite quadrature (GHQ) approximation can be applied but is only suitable for low-dimensional random effects. Based on the Quasi-Monte Carlo (QMC) approximation, a heuristic approach is proposed to calculate the maximum likelihood estimates of parameters in the GLMM. The QMC points scattered uniformly on the high-dimensional integration domain are generated to replace the GHQ nodes. Compared to the GHQ approximation, the proposed method has many advantages such as its affordable computation, good approximation and fast convergence. Comparisons to the penalized quasi-likelihood estimation and Gibbs sampling are made using a real dataset and a simulation study. The real dataset is the salamander mating dataset whose modelling involves six 20-dimensional intractable integrals in the likelihood.     

Recurrent type-2 fuzzy neural network using Haar wavelet energy and entropy features for speech detection in noisy environments

This paper proposes a new method to detect the boundary of speech in noisy environments. This detection method uses Haar wavelet energy and entropy (HWEE) as detection features. The Haar wavelet energy (HWE) is derived by using the robust band that shows the most significant difference between speech and nonspeech segments at different noise levels. Similarly, the wavelet energy entropy (WEE) is computed by selecting the two wavelet energy bands whose entropy shows the most significant speech/nonspeech difference. The HWEE features are fed as inputs to a recurrent self-evolving interval type-2 fuzzy neural network (RSEIT2FNN) for classification. The RSEIT2FNN is used because it uses type-2 fuzzy sets, which are more robust to noise than type-1 fuzzy sets. The recurrent structure in the RSEIT2FNN helps to remember the context information of a test frame. The RSEIT2FNN outputs are compared with a parameter threshold to determine whether it is a speech or nonspeech period. The HWEE-based RSEIT2FNN detection was applied to speech detection in different noisy environments with different noise levels. Comparisons with different detection methods verified the advantage of the proposed method of using HWEE.     

Improved Voice Activity Detection Using Contextual Multiple Hypothesis Testing for Robust Speech Recognition

This paper shows an improved statistical test for voice activity detection in noise adverse environments. The method is based on a revised contextual likelihood ratio test (LRT) defined over a multiple observation window. The motivations for revising the original multiple observation LRT (MO-LRT) are found in its artificially added hangover mechanism that exhibits an incorrect behavior under different signal-to-noise ratio (SNR) conditions. The new approach defines a maximum a posteriori (MAP) statistical test in which all the global hypotheses on the multiple observation window containing up to one speech-to-nonspeech or nonspeech-to-speech transitions are considered. Thus, the implicit hangover mechanism artificially added by the original method was not found in the revised method so its design can be further improved. With these and other innovations, the proposed method showed a higher speech/nonspeech discrimination accuracy over a wide range of SNR conditions when compared to the original MO-LRT voice activity detector (VAD). Experiments conducted on the AURORA databases and tasks showed that the revised method yields significant improvements in speech recognition performance over standardized VADs such as ITU T G.729 and ETSI AMR for discontinuous voice transmission and the ETSI AFE for distributed speech recognition (DSR), as well as over recently reported methods.     

Adaptive Gaussian filters for nonlinear state estimation with one-step randomly delayed measurements

This paper proposes new algorithms of adaptive Gaussian filters for nonlinear state estimation with maximum one-step randomly delayed measurements. The unknown random delay is modeled as a Bernoulli random variable with the latency probability known a priori. However, a contingent situation has been considered in this work when the measurement noise statistics remain partially unknown. Due to unavailability of the complete knowledge of measurement noise statistics, the unknown measurement noise covariance matrix is estimated along with states following: (i) variational Bayesian approach, (ii) maximum likelihood estimation. The adaptation algorithms are mathematically derived following both of the above approaches. Subsequently, a general framework for adaptive Gaussian filter is presented with which variants of adaptive nonlinear filters can be formulated using different rules of numerical approximation for Gaussian integrals. This paper presents a few of such filters, viz., adaptive cubature Kalman filter, adaptive cubature quadrature Kalman filter with their higher degree variants, adaptive unscented Kalman filter, and adaptive Gauss–Hermite filter, and demonstrates the comparative performance analysis with the help of a nontrivial Bearing only tracking problem in simulation. Additionally, the paper carries out relative performance comparison between maximum likelihood estimation and variational Bayesian approaches for adaptation using Monte Carlo simulation. The proposed algorithms are also validated with the help of an off-line harmonics estimation problem with real data.     

Tracking of Nonstationary Noise Based on Data-Driven Recursive Noise Power Estimation

This paper considers estimation of the noise spectral variance from speech signals contaminated by highly nonstationary noise sources. The method can accurately track fast changes in noise power level (up to about 10 dB/s). In each time frame, for each frequency bin, the noise variance estimate is updated recursively with the minimum mean-square error (mmse) estimate of the current noise power. A time- and frequency-dependent smoothing parameter is used, which is varied according to an estimate of speech presence probability. In this way, the amount of speech power leaking into the noise estimates is kept low. For the estimation of the noise power, a spectral gain function is used, which is found by an iterative data-driven training method. The proposed noise tracking method is tested on various stationary and nonstationary noise sources, for a wide range of signal-to-noise ratios, and compared with two state-of-the-art methods. When used in a speech enhancement system, improvements in segmental signal-to-noise ratio of more than 1 dB can be obtained for the most nonstationary noise sources at high noise levels.     

改进相位谱补偿的语音增强方法

针对传统单通道语音增强方法中用带噪语音相位代替纯净语音相位重建时域信号，使得语音主观感知质量改善受限的情况，提出了一种改进相位谱补偿的语音增强算法。该算法提出了基于每帧语音输入信噪比的Sigmoid型相位谱补偿函数，能够根据噪声的变化来灵活地对带噪语音的相位谱进行补偿；结合改进DD的先验信噪比估计与语音存在概率算法（SPP）来估计噪声功率谱；在维纳滤波中结合新的语音存在概率噪声功率谱估计与相位谱补偿来提高语音的增强效果。相比传统相位谱补偿（PSC）算法而言，改进算法可以有效抑制音频信号中的各类噪声，同时增强语音信号感知质量，提升语音的可懂度。     

飞机舱音信息鲁棒语音端点检测

提出了一种适用于飞机座舱噪声环境下的鲁棒语音端点检测方法.在分析噪声特征的基础上,首先直接针对带噪语音谱的离散傅里叶变换系数建立复拉普拉斯分布模型;然后进行基于二元假设检验的似然比测试;最后将信号相邻帧的相关性与基于最大后验概率的判决规则相结合.定义两种门限值,根据前一帧的状态与当前帧的观测值共同决策当前帧的状态,从而搜索出语音起止点.实验表明:与目前典型语音端点检测算法对比,该方法在飞机座舱噪声环境中具有较好的鲁棒性.     

非平稳噪声环境下的噪声功率谱估计方法

提出了一种新的非平稳噪声环境下的噪声功率谱估计方法。该方法通过采用非固定长度的时间窗跟踪含噪语音功率谱的最小值,解决了传统跟踪时延较大的问题。不同频带采用不同的阈值计算语音存在概率,从而利用语音存在概率值的大小调节噪声和语音的混合程度。实验证明,本文提出的方法较基于语音活性判决(Voiceactivity detectors,VAD)的一系列方法和传统的最小统计(Minimal statistic,MS)算法有更好的效果,从而有效地改善了增强后语音的质量。