一种改进的卡尔曼滤波语音增强法

提出一种基于谱减法和听觉掩蔽效应的改进的卡尔曼滤波语音增强算法.引入基于谱减法的AR参数估计使卡尔曼算法降低了复杂度和计算量从而易于实现.用卡尔曼滤波滤除噪声的同时结合人耳听觉掩蔽特性设计一个后置感知滤波器,使得从卡尔曼滤波获得的估计误差低于人耳掩蔽阈值,在去噪和语音失真之间取较好的折中.仿真结果表明所提方法优于传统的卡尔曼滤波增强法,能够有效地减少语音失真,并且更符合人耳听觉特性,特别是在低信噪比的情况下,语音具有更好的清晰度和可懂度.     

基于多统计模型和人耳听觉特性的麦克风阵列后滤波语音增强算法

针对麦克风阵列后滤波语音增强算法的不足, 结合人耳的听觉掩蔽效应, 提出了改进的后滤波语音增强算法. 提出了最大化目标语音存在概率来确定信号子空间维度的方法, 在噪声子空间上, 利用条件概率估计出噪声功率谱. 基于人耳的听觉掩蔽效应, 提出了后滤波器的一种合理的设计方法. 实验证明, 所提的噪声估计方法比传统方法更加准确, 所提的后滤波算法比传统的后滤波算法更好, 在多项语音评价指标上, 都取得了更好的实验效果.     

基于感知掩蔽深度神经网络的单通道语音增强方法

本文将心理声学掩蔽特性应用于基于深度神经网络（Deep neural network,DNN）的单通道语音增强任务中,提出了一种具有感知掩蔽特性的DNN结构.首先,提出的DNN对带噪语音幅度谱特征进行训练并分别得到纯净语音和噪声的幅度谱估计.其次,利用估计的纯净语音幅度谱计算噪声掩蔽阈值.然后,将噪声掩蔽阈值和估计的噪声幅度谱联合计算得到一个感知增益函数.最后,利用感知增益函数从带噪语音幅度谱中估计出增强语音幅度谱.在TIMIT数据库上,对不同信噪比下的20种噪声进行的仿真实验表明,无论噪声类型是否在语音的训练集中出现,所提出的感知掩蔽DNN都能够在有效去除噪声的同时保持较小的语音失真,增强效果明显优于常见的DNN增强方法以及NMF（Nonnegative matrix factorization）增强方法.     

Robust speech recognition system using multi-parameter bidirectional Kalman filter

A speech signal processing system using multi-parameter model bidirectional Kalman filter has been proposed in this paper. Conventional unidirectional Kalman filter usually performs estimation of current state speech signal by processing the time varying autoregressive model of speech signals from the past time states. A bidirectional Kalman filter utilizes the past and future measurements to estimate the current state of a speech signal that minimize the mean of the squared error using efficient recursive means. The matrices involved in the difference equations and the measurement equations of the bidirectional Kalman filter algorithm are kept constant throughout the process. With multi-parameter model, the proposed bidirectional Kalman filter relates more measurements from the future and past time states to the current time state. The proposed multi-parameter bidirectional Kalman filter has been implemented into a speech recognition system and its performance has been compared to other conventional speech processing algorithms. Compared to the single-parameter model bidirectional Kalman filter, the multi-parameter bidirectional Kalman filter improves the accuracy in the state prediction, reduces the speech information lost after the filtering process and better word error rate has been achieved at high SNR regions (clean, 20, 15, 10 dB).     

稳键的自适应空间波束形成语音增强算法研究

为了进一步改善波束形成的降噪性能,研究了一种稳键后置滤波自适应空间波束形成算法.用麦克风代替传统波束形成器的延时抽头线,使所有的麦克风都有一阶的滤波器,利用经典的线性约束最小方差准则使空间波束形成产生语音参考信号,同阻塞矩阵输出的噪声参考信号一起经自适应多路相消器,从而有效的消除干扰噪声;最后结合后置滤波技术进一步改善语音质量.实验结果表明,相对于传统后置滤波自适应波束形成算法,在消噪性能上有明显的改善且具有更高的输出信噪比.     

一种听觉掩蔽效应和维纳滤波的语音增强算法

将非平稳噪声估计算法以及基于听觉掩蔽效应得到的噪声被掩蔽概率应用于维纳滤波语音增强中,提出了一种听觉掩蔽效应和维纳滤波的语音增强方法。几种噪声背景下对语音增强的客观测试表明,提出的算法相比较于传统的维纳滤波语音增强算法而言不但可以提高语音信噪比,而且可以明显减少语音失真。     

A generalized time-frequency subtraction method for robust speech enhancement based on wavelet filter banks modeling of human auditory system.

We present a new speech enhancement scheme for a single-microphone system to meet the demand for quality noise reduction algorithms capable of operating at a very low signal-to-noise ratio. A psychoacoustic model is incorporated into the generalized perceptual wavelet denoising method to reduce the residual noise and improve the intelligibility of speech. The proposed method is a generalized time-frequency subtraction algorithm, which advantageously exploits the wavelet multirate signal representation to preserve the critical transient information. Simultaneous masking and temporal masking of the human auditory system are modeled by the perceptual wavelet packet transform via the frequency and temporal localization of speech components. The wavelet coefficients are used to calculate the Bark spreading energy and temporal spreading energy, from which a time-frequency masking threshold is deduced to adaptively adjust the subtraction parameters of the proposed method. An unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. Through rigorous objective and subjective evaluations, it is shown that the proposed speech enhancement system is capable of reducing noise with little speech degradation in adverse noise environments and the overall performance is superior to several competitive methods.     

A Generalized Time–Frequency Subtraction Method for Robust Speech Enhancement Based on Wavelet Filter Banks Modeling of Human Auditory System

We present a new speech enhancement scheme for a single-microphone system to meet the demand for quality noise reduction algorithms capable of operating at a very low signal-to-noise ratio. A psychoacoustic model is incorporated into the generalized perceptual wavelet denoising method to reduce the residual noise and improve the intelligibility of speech. The proposed method is a generalized time-frequency subtraction algorithm, which advantageously exploits the wavelet multirate signal representation to preserve the critical transient information. Simultaneous masking and temporal masking of the human auditory system are modeled by the perceptual wavelet packet transform via the frequency and temporal localization of speech components. The wavelet coefficients are used to calculate the Bark spreading energy and temporal spreading energy, from which a time-frequency masking threshold is deduced to adaptively adjust the subtraction parameters of the proposed method. An unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. Through rigorous objective and subjective evaluations, it is shown that the proposed speech enhancement system is capable of reducing noise with little speech degradation in adverse noise environments and the overall performance is superior to several competitive methods.     

基于小波变换和Kalman滤波的语音增强方法

针对受加性噪声干扰的语音信号,采用基于小波变换的Kalman滤波方法,提出一种有效的语音增强方法.分析在实际处理中所遇到的二进小波变换、滤波参数估计、Kalman滤波发散等问题.语音增强的效果采用信噪比来进行评估.仿真实验表明在加性噪声为高斯白噪声和色噪的情况下,该方法均具有较好的有效性.     

基于噪声估计的二值掩蔽语音增强算法

针对现有的助听器语音增强算法在非平稳噪声环境下,残留大量背景噪声的同时还引入了“音乐噪声”,致使增强语音可懂度和信噪比不理想等问题。提出了一种基于噪声估计的二值掩蔽语音增强算法,该算法利用人耳听觉感知理论,结合人耳的听觉特性和耳蜗的工作机理。采用最小值控制递归平均（Minima-Controlled Recursive Averaging,MCRA）算法获得估计噪声和初步增强语音;将估计噪声和初步增强语音分别通过可以模拟人工耳蜗模型的gammatone滤波器组进行滤波处理,得到各自的时频表示形式;利用人耳的听觉掩蔽特性,计算含噪语音在时频域的二值掩蔽;利用二值掩蔽得到增强语音。实验结果表明：该算法很大程度上去除了谱减法引入的“音乐噪声”,与基于MCRA谱减法相比,增强语音的语言可懂度指数（Speech Intelligibility Index,SII）、主观语音质量评估（Perceptual Evaluation of Speech Quality,PESQ）和信噪比（Signal to Noise Ratio,SNR）都得到了提高。     

混合式自适应Kalman滤波算法

采用虚拟噪声补偿模型误差和有偏的噪声方差估值器、滤波器收敛性判据相结合的方法来解决自适应Kalman滤波发散的问题。首先若模型不准确,则引入虚拟噪声对模型误差进行虚拟补偿,然后采用有偏的噪声方差估值器、滤波器收敛性判据对噪声方差估计值进行监控,阻止滤波器发散。采用混合式自适应Kalman滤波算法对Gill公司的风向风速仪实时采集的数据进行处理,实验结果表明,该方法能有效的提高性能、抑制滤波发散,具有较强的实用性、自适应能力。     

On the choice of importance distributions for unconstrained and constrained state estimation using particle filter

Recursive state estimation of constrained nonlinear dynamical system has attracted the attention of many researchers in recent years. For nonlinear/non-Gaussian state estimation problems, particle filters have been widely used (Arulampalam et al. [1]). As pointed out by Daum [2], particle filters require a proposal distribution and the choice of proposal distribution is the key design issue. In this paper, a novel approach for generating the proposal distribution based on a constrained Extended Kalman filter (C-EKF), Constrained Unscented Kalman filter (C-UKF) and constrained Ensemble Kalman filter (C-EnkF) has been proposed. The efficacy of the proposed state estimation algorithms using a particle filter is illustrated via a successful implementation on a simulated gas-phase reactor, involving constraints on estimated state variables and another example problem, which involves constraints on the process noise (Rao et al. [10]). We also propose a state estimation scheme for estimating state variables in an autonomous hybrid system using particle filter with Unscented Kalman filter as a proposal and unconstrained Ensemble Kalman filter (EnKF) as a proposal. The efficacy of the proposed state estimation scheme for an autonomous hybrid system is demonstrated by conducting simulation studies on a three-tank hybrid system. The simulation studies underline the crucial role played by the choice of proposal distribution in formulation of particle filters.     

Digital watermarking algorithm based on Kalman filtering and image fusion

A digital watermarking algorithm based on Kalman filter and image fusion is proposed. The digital watermarking can be viewed as a process that embedding a weak signal (watermark) to a strong signal (original image), so the process of watermarking can be viewed as a process of image fusion. In the proposed watermarking algorithm, the watermark embedding and extraction process are expressed as the state estimate process, and Kalman filter is used as an optimal estimation algorithm in the process of image fusion. An optimal estimation model is built according to the watermark image and the original image, and then the state equation and the corresponding measurement equation are built. The optimal estimation is archived in case of the minimum estimation error variance. Crossentropy and mutual information are used to evaluate the performance of image fusion. Experimental results show that the proposed algorithm has a good performance in both robustness and invisibility.     

基于TF-GSC的多通道后置滤波语音增强算法

针对传统语音增强算法在非平稳噪声,尤其是在噪声为语音的环境下,对噪声的抑制效果急剧下降的情况,提出了一种基于传递函数—广义旁瓣抵消(TF-GSC)和最佳修正测井谱振幅估计量(OM-LSA)的改进型多通道后置滤波语音增强算法.算法在后置滤波时,利用TF-GSC输出信号与参考噪声之间的相互关系求解出语音存在概率,并更新噪声功率谱估计.实验结果表明:算法可以有效地抑制非平稳噪声,提高语音增强算法在语音噪声环境下的鲁棒性.     

On the properties of reduced order Kalman filters

Several known results are unified by considering properties of reduced-order Kalman filters. For the case in which the number of noise sources equals the number of observations, it is shown that the reduced-order Kalman filter achieves zero steady-state variance of the estimation error if and only if the plant has no transmission zeros in the right-half plane, since these would be among the poles of the Kalman filter. The reduced-order Kalman filter cannot achieve zero variance of the estimation error if the number of independent noise sources exceeds the number of observations. It is also shown that the reduced-order Kalman filter achieves the generalized Doyle-Stein condition for robustness when the noise sources are colocated with the control inputs. When there are more observations than noise sources, additional noise sources can be postulated to improve the observer frequency response without diminishing robustness     

Mobile robot localization with gyroscope and constrained Kalman filter

The odometry information used in mobile robot localization can contain a significant number of errors when robot experiences slippage. To offset the presence of these errors, the use of a low-cost gyroscope in conjunction with Kalman filtering methods has been considered by many researchers. However, results from conventional Kalman filtering methods that use a gyroscope with odometry can unfeasible because the parameters are estimated regardless of the physical constraints of the robot. In this paper, a novel constrained Kalman filtering method is proposed that estimates the parameters under the physical constraints using a general constrained optimization technique. The state observability is improved by additional state variables and the accuracy is also improved through the use of a nonapproximated Kalman filter design. Experimental results show that the proposed method effectively offsets the localization error while yielding feasible parameter estimation.     

Audible Noise Reduction in Eigendomain for Speech Enhancement

A signal subspace scheme based on masking properties is proposed for enhancement of speech degraded by additive noise. Since the masking properties are related to the critical frequency band that is derived from the characteristics of human cochlea, the incorporation of masking threshold into a subspace technique requires the transformation between the frequency and eigen domains. We present and apply an invertible transformation between the frequency and eigen domains. In this paper, we use masking properties of the human auditory system to define the audible noise quantity in the eigendomain. We derive the eigen-decomposition of the estimated speech autocorrelation matrix with the assumption of white noise. Subsequently, an audible noise reduction scheme is developed based on a signal subspace technique, and the implementation of our proposed scheme is outlined. We further extend the scheme to the colored noise case. Simulation results show the superiority of our proposed scheme over other existing subspace methods in terms of segmental signal-to-noise ratio (SNR), perceptual evaluation of speech quality (PESQ), modified Bark spectral distortion (MBSD), spectrogram and informal listening tests.     

脉冲噪声环境下基于卡尔曼滤波的语音增强

语音增强的目标在于从含噪信号中提取纯净语音,纯净语音在某些环境下会被脉冲噪声所污染,但脉冲噪声的时域分布特征却给语音增强带来困难,使传统方法在脉冲噪声环境下难以取得满意效果。为在平稳脉冲噪声环境下进行语音增强,提出了一种新方法。该方法通过计算确定脉冲噪声样本的能量与含噪信号样本的能量之比最大的频段,利用该频段能量分布情况逐帧判别语音信号是否被脉冲噪声所污染。进一步地,该方法只在被脉冲噪声污染的帧应用卡尔曼滤波算法去噪,并改进了传统算法执行时的自回归(AR)模型参数估计过程。实验中,采用白色脉冲噪声以及有色脉冲噪声污染语音信号,并对低输入信噪比的信号进行语音增强,结果表明所提出的算法能显著地改善信噪比和抑制脉冲噪声。     

Robust unscented Kalman filter for nanosat attitude estimation

A robust unscented Kalman filter based on a multiplicative quaternion-error approach is proposed for nanosat estimation in the presence of measurement faults. The global attitude parameterization is given by a quaternion, while the local attitude error is defined using a generalized three-dimensional attitude representation. The proposed algorithm uses a statistical function including measurement residuals to detect measurement faults and then uses an adaptation scheme based on multiple measurement scale factor for filter robustness against faulty measurements. The proposed algorithm is demonstrated for the attitude estimation of a nanosat with an on-board three-axis magnetometer and rate-integrating gyros in the presence of measurement faults as well as satellite orbit errors. To compare the estimation performance of the proposed algorithm, the robust unscented Kalman filter with single measurement noise scale factor, the standard extended Kalman filter and the unscented Kalman filter are also implemented under the same simulation conditions.     

基于转换量测容积卡尔曼滤波器带多普勒量测的目标跟踪算法

针对带多普勒量测的目标跟踪问题,提出一种基于转换量测容积卡尔曼滤波器的序贯滤波目标跟踪算法.对具有量测误差相关性的距离和多普勒量测进行解相关处理,构造出新的解相关量测方程,进而基于贝叶斯方法提出带多普勒量测的序贯处理算法的统一理论框架,实现对位置量测和多普勒量测的序贯滤波.在该理论框架下,提出基于转换量测容积卡尔曼滤波器的序贯滤波目标跟踪算法.该算法先采用转换量测容积卡尔曼滤波器和位置量测对目标状态进行估计,再利用经典容积卡尔曼滤波器对新构造的伪多普勒量测进行量测更新以实现目标跟踪.通过对所提算法的性能分析验证该算法的一致性和收敛性.仿真结果表明,该算法与其他跟踪算法相比,具有更高的跟踪精度.