应对说话人位置突变的鲁棒语声去混响*

多通道线性预测算法是一种有效的语声去混响方法,但若房间内说话人位置发生变化,房间冲激响应也会随之改变,从而导致去混响效果变差。本文研究了基于递归最小二乘算法的自适应多通道线性预测语声去混响方法,提出了一种去混响过程中检测说话人位置突变的方法,它借助已有的多通道传声器信号,并未明显增加系统开销。实验表明该方法可以有效检测说话人位置的突变,并通过重置滤波器改善算法的收敛速度和稳定性。     

偏度最大化多通道逆滤波语音去混响研究*

房间混响会降低语音质量和语音可懂度。高阶统计量是衡量非高斯性的重要参量,基于语音非高斯特性可实现语音去混响。本文提出一种基于高阶统计量的多通道语音去混响方法,该方法首次用多通道语音信号线性预测残差的三阶统计量偏度（Skewness）构造代价函数,以去混响重建信号线性预测残差的偏度最大化为目标自适应地更新逆滤波器;同时结合语音信号的产生模型,提出基于偏度准则的线性预测与房间脉冲响应逆滤波联合估计方法,进一步提高去混响算法性能。实验结果表明,该方法相较于已有的基于线性预测残差四阶统计量峰度（Kurtosis）的方法具有更好的去混响效果,且对噪声具有更强的鲁棒性。     

球谐域自适应混响抵消与声源定位算法

提出了一种基于球谐域的自适应混响抵消与声源定位算法,该方法通过去混响处理改善语音质量,并提高球谐域定位算法在混响环境下的定位性能。推导了基于多通道线性预测的自适应混响抵消算法在球谐域的表达式,针对刚球模型提出分阶处理的去混响方法,并对去混响后的信号进行波达方向估计。采用32元球阵的仿真结果表明,相比于球谐域不分阶去混响方法,该方法最大可减少约2/3的运算量,同时语音PESQ得分及SRMR均显著提高。利用实验数据对算法性能进行测试,实验结果验证了该方法在实际声学环境中去混响和声源定位的有效性。      

分布式无线声传感网加权预测误差语声去混响方法

室内混响会严重降低语声质量,因此在室内语声通信中对混响的抑制显得尤为重要.针对无线声传感网,该文提出一种基于加权预测误差的分布式自适应去混响算法.通过调整传统递归最小二乘算法,所提出的分布式加权预测误差算法仅需利用一路相同的参考信号和其他节点的本地输出而非全部信号,便可实现最优输出,从而大幅度降低节点间传输的通道数与各...     

有源头靠中一种分布式虚拟传声器优化方法

有源降噪头靠系统中,远程虚拟传声器技术能够解决控制点处与误差传声器处降噪量不匹配的问题。在实际应用中,多通道虚拟传声器技术存在收敛速度慢和运算复杂度高等问题。针对这个问题,本文通过重新设计远程虚拟传声技术的离线优化过程,提出一种分布式远程虚拟传声器技术优化方法。该方法将虚拟次级通路矩阵作对角化限制,同时对观测传递函数矩阵进行联合寻优,以实现一种分布式的更新算法。有源降噪头靠实验结果表明,所提算法能够有效降低远程虚拟传声器技术算法的运算复杂度,并且提升了算法的收敛速度。     

偏度最大化多通道逆滤波语声去混响研究

房间混响会降低语声质量和语声可懂度。高阶统计量是衡量非高斯性的重要参量,基于语声非高斯特性可实现语声去混响。该文提出一种基于高阶统计量的多通道语声去混响方法,该方法首次用多通道语声信号线性预测残差的三阶统计量偏度构造代价函数,以去混响重建信号线性预测残差的偏度最大化为目标自适应地更新逆滤波器,同时引入通道逆滤波和语声产生系统的联合估计。实验结果表明,该方法相较于已有的基于线性预测残差四阶统计量峰度的方法具有更好的去混响效果,且对噪声具有更强的鲁棒性。     

两阶段复数谱卷积循环网络立体声回声消除

提出了一种两阶段复数谱卷积循环网络(CRN)的立体声回声消除(SAEC)算法,该算法无需对立体声信号进行去相关,因而能够在保证立体声音质和空间感的同时,解决自适应滤波SAEC算法非唯一解问题。所提算法采用两个阶段进行回声消除,第一阶段根据传声器接收信号和参考信号估计回声信号,第二阶段将估计回声信号作为先验信息,联合传声器接收信号作为输入特征,估计近端语音。相对于单阶段CRN算法,该方法能够提高网络对回声和近端语音的区分度,有助于近端语音的提取。另外,网络的输入特征和训练目标均采用复数谱,降低了近端语音的相位估计误差,因而可以进一步提升算法性能。实验表明,基于两阶段复数谱CRN的SAEC算法在单端讲话时的回声抑制量和双端讲话时的语音质量都明显优于传统算法以及单阶段CRN算法。      

联合宽带到达方向估计和语音特征增强的传声器阵处理方法

以提高室内混响环境下自动语音识别(ASR)性能为目标,讨论了一种小尺寸传声器阵处理方法.该方法采用基于旋转不变技术的信号参数估计算法(ESPRIT)计算宽带语音信号到达方向角,进行时延补偿;同时联合考虑阵列滤波与隐马尔可夫模型(HMM)识别过程,将识别输出结果反馈到前端的传声器阵处理,优化阵列滤波系数.与常规阵处理方法改善信号波形质量不同,本文通过调节阵列滤波系数降低待识别特征与训练模型之间的失配,直接提高识别过程中正确假设的概率.实验结果表明,上述方案能够有效降低会议室环境下孤立词有限词库ASR的错误概率,表现优于常规波束形成方法;采用全局优化进行阵列滤波设计,与局部优化算法相比,进一步改善了处理性能.     

低复杂度的盲源分离和去混响联合优化方法

提出了一种低复杂度的基于加权预测误差(WPE)的独立低秩矩阵分析(ILRMA)方法。与现有的WPE-ILRMA方法把预测矩阵当成一个整体来处理不同,所提方法将预测矩阵展开来推导代价函数,利用不同声源的混合滤波器和分离滤波器之间的正交性简化代价函数的优化过程,进而以更低的计算复杂度对混合信号去混响。通过利用解耦预测矩阵和分离滤波器之间的关系,所提方法将维数较大的矩阵求逆转化为维数较小的矩阵求逆,从而取得了比WPE-ILMRA方法更低的计算复杂度。在最大似然框架下推导了所提方法的代价函数,并采用坐标梯度下降算法来估计参数。实验结果表明,所提方法能以更低的计算复杂度和更高的稳定性取得与WPE-ILRMA方法相似的分离性能。     

基于二次相关的语音信号时延估计改进算法

目前语音信号的时延估计研究,大部分采用的是广义互相关算法。然而,广义互相关时延估计算法易受噪声和混响环境影响。为此,本文提出了一种基于二次相关的语音信号时延估计改进算法,该算法对语音信号进行二次互相关运算,并结合Hilbert变换,对二次互相关峰值进行进一步的锐化处理,使得反映时延的峰值点检测更为准确。实验结果表明,改进的时延估计方法在非平稳的语音信号中能够有效地抑制噪声干扰,且在不同混响条件下时延估计具有更好的性能。     

Single-channel dereverberation using a non-causal minimum variance distortionless response filter

This letter presents a single-channel speech dereverberation approach using a non-causal minimum variance distortionless response (MVDR) filter. The non-causal filter is adopted to utilize the additional information of the desired signal that lies in subsequent frames. Note that the desired signal output has minimal distortion due to the introduction of the MVDR criterion. The proposed system further suppresses the late reverberation by employing a statistical reverberant model. Experimental results demonstrate the superiority of the proposed algorithm to conventional approaches.     

扩散噪声下协方差矩阵重构的语音分离与降噪

针对传统多通道语音分离算法在扩散噪声下性能下降的问题,提出了一种用于语音分离及降噪的空间协方差模型及参数估计方法。该方法将扩散噪声视为独立声源,利用由导向矢量重构的空间协方差矩阵建模目标声源的空间特性,并通过空间协方差分析方法估计用于语音分离的多通道维纳滤波器。同时,还提出了一种联合该方法的后置滤波器参数框架,为输出信号降噪和失真的折中提供了更多选择。在扩散噪声下的单目标和多目标实验中,所提方法的语音提取和分离性能都优于对比算法,联合参数的后置滤波器可提供更为符合人们要求的降噪语音,验证了所提模型与参数估计方法的有效性。      

Wide-band dereverberation method based on multichannel linear prediction using prewhitening filter

Several dereverberation algorithms have been studied. The sampling frequencies used in conventional studies are typically 8–16 kHz because their main purpose is preprocessing for improving the intelligibility of speech communication and articulation for automatic speech recognition. However, in next-generation communication systems, techniques to analyze and reproduce not only semantic information of sound but also more high-definition components such as spatial information and directivity will be increasingly necessary. To decompose these sound field characteristics with high definition, a dereverberation algorithm that is useful at high sampling frequencies is an important technique to process sound that includes high-frequency spectra such as musical sounds. The LInear-predictive Multichannel Equalization (LIME) algorithm is a promising dereverberation method. Using the LIME algorithm, however, a dereverberation signal cannot be solved at high sampling frequencies when the source signal is colored, such as in the case of speech and sound of musical signals. Because the rank of the correlation matrix calculated from such a colored signal is not full, the characteristic polynomial cannot be calculated precisely. To alleviate this problem, we propose preprocessing of all input signals with filters to whiten their spectra so that this algorithm can function for colored signals at high sampling frequencies.     

MR fingerprinting reconstruction with Kalman filter

Magnetic resonance fingerprinting (MR fingerprinting or MRF) is a newly introduced quantitative magnetic resonance imaging technique, which enables simultaneous multi-parameter mapping in a single acquisition with improved time efficiency. The current MRF reconstruction method is based on dictionary matching, which may be limited by the discrete and finite nature of the dictionary and the computational cost associated with dictionary construction, storage and matching.In this paper, we describe a reconstruction method based on Kalman filter for MRF, which avoids the use of dictionary to obtain continuous MR parameter measurements. With this Kalman filter framework, the Bloch equation of inversion-recovery balanced steady state free-precession (IR-bSSFP) MRF sequence was derived to predict signal evolution, and acquired signal was entered to update the prediction. The algorithm can gradually estimate the accurate MR parameters during the recursive calculation. Single pixel and numeric brain phantom simulation were implemented with Kalman filter and the results were compared with those from dictionary matching reconstruction algorithm to demonstrate the feasibility and assess the performance of Kalman filter algorithm.The results demonstrated that Kalman filter algorithm is applicable for MRF reconstruction, eliminating the need for a pre-define dictionary and obtaining continuous MR parameter in contrast to the dictionary matching algorithm.     

混沌信号在无线传感器网络中的盲分离

混沌信号在本质上属于非线性非高斯信号,它在无线传感器网络下的应用还涉及到信号量化问题,这使得混沌信号在此应用环境下的信号盲分离更为棘手.针对此问题,本文在容积卡尔曼粒子滤波的框架下提出一种解决方法.文中首先推导出观测信号的概率密度函数,在量化比特有限的情况下,采用最优量化器,获得最优的量化结果.在此基础上,使用容积卡尔曼滤波器产生粒子滤波中的重要性概率密度函数,融入最新的观测值,提高粒子对系统状态后验概率的逼近,提高信号盲分离的精度.仿真结果表明算法能够有效地分离混合混沌信号,参数估计的精度及其运算量均优于已有的无先导卡尔曼粒子滤波算法,其运行时间为无先导卡尔曼粒子滤波算法的88.77%.     

An innovation based random weighting estimation mechanism for denoising fiber optic gyro drift signal

In Interferometric Fiber Optic Gyroscope (IFOG), the diminution of random noise and drift error is a critical task. These errors degrade the performance of IFOG. In this paper, a modified adaptive Kalman gain correction (AKFG) algorithm is proposed to denoise IFOG signal. The covariance matrix of innovation sequence is estimated using weighted average window method in which the weights are randomly generated in the range [0, 1]. Innovation based random weighted estimation (IRWE)-AKFG is applied to denoise the IFOG drift signal. The Kalman gain is adaptively updated using the covariance matrix of innovation sequence. The proposed algorithm is applied for denoising IFOG signal under static and dynamic environment. Allan variance method is used to analyze and quantify the stochastic errors in IFOG sensor. The performance of the proposed algorithm is compared with Conventional Kalman filter (CKF) and the simulation results reveal that the proposed algorithm is an efficient algorithm for denoising the IFOG signal.     

一种无线传感器网络中的混沌信号重构算法

将无线传感器网络节点观测区域中的一个混沌信号发送到融合中心,进行信号重构.由于节点的通信带宽受限,信号传输之前需要进行量化,给信号带来量化噪声,使得信号重构工作变得更为棘手.本文提出用平方根容积卡尔曼滤波器对融合中心收集的信号进行重构.首先估计观测信号的概率密度函数,使用最优量化器量化观测信号,在有限的量化比特数下,取得最优的信号量化性能.平方根容积卡尔曼滤波器相对无先导卡尔曼算法具有较少的求容积分点,因此具有计算量小的优点,同时迭代过程采用传递误差矩阵的平方根矩阵,保证迭代过程的稳定性和提高数据估计精度.仿真结果表明,该算法能够有效和快速地重构观测信号,并且比基于无先导卡尔曼滤波的算法更快.