基于频率信噪比加权的麦克风阵列声源定位算法

为了提高噪声和混响环境下麦克风阵列的声源定位算法性能,提出了一种基于频率信噪比加权的可控响应功率定位算法。该算法首先根据每帧阵列信号的频域协方差矩阵估计每个频率的信噪比;然后通过激活函数将频率信噪比映射为加权值,并修正传统的相位变换可控响应功率计算公式;最后利用修正公式计算每个候选位置的可控响应功率值,通过搜索可控响应功率的最大值实现声源定位。该算法根据实时估计的频率信噪比自适应地调整各频率分量对可控响应功率的贡献。仿真结果表明,与传统的相位变换可控响应功率算法、维纳预滤波波束形成算法相比,在噪声和混响的复杂声学环境下,本文算法的定位正确率更高,均方根误差更小,对噪声的鲁棒性更强。      

基于分层采样粒子滤波的麦克风阵列说话人跟踪方法

针对噪声与混响环境下的说话人跟踪问题,本文提出了一种基于粒子滤波的麦克风阵列声源定位与跟踪方法.该方法在粒子滤波框架下,将无混响影响的语音建立信号作为观测信息,通过计算麦克风阵列波束形成器的输出能量来构建似然函数,同时考虑语音信号不同频率成分在声源定位中的作用,利用分层采样方法提高粒子的采样效率.实验结果表明,本文方法提高了说话人声源跟踪系统的抗噪声与抗混响能力.     

复杂环境下基于时延估计的声源定位技术研究简

为了改善在复杂环境下声源定位算法的性能,提出了一种新的时延估计（TDE）方法,即基于传递函数比的统计模型方法（ATFR-SM）。该方法采用统计模型去除噪声对传递函数（ATF）的影响,在计算传递函数时对功率谱密度（PSD）进行平滑和“白化”,以去除混响对传递函数的影响。同时,算法中引入话音激活检测（VAD）去除对求取传递函数无用的噪声段,以提高时延估计的准确性。此外,将所提时延估计方法与线性定位法相结合,构成一套完整的声源定位方法。实验结果表明,在复杂环境下,时延估计方法具有更低的异常点百分比（PAP）和均方根误差（RMSE）,且明显优于传统的参考算法,同时声源定位方法具有更高的定位精度。     

Robust Voice Activity Detection Based on Concept of Modulation Transfer Function in Noisy Reverberant Environments

Voice activity detection (VAD) is used to detect speech and non-speech periods from observed speech signals. It is an important front-end technique for many speech technology applications. Many VAD methods have been proposed. However most of them have been applied under clean or noisy conditions. Only a few methods have been proposed for reverberant conditions, particularly under noisy reverberant conditions. We therefore need to understand the ill effects of noise and reverberation on speech to design an accurate and robust method of VAD under noisy reverberant conditions. The ill effects of noise and reverberation for speech can be regarded as the modulation transfer function (MTF) under noisy and reverberant conditions. Therefore, our study is based on the MTF concept to reduce the ill effects of noise and reverberation on speech, and propose a robust VAD method that we obtained in this study. Noise reduction and dereverberation were first applied to the temporal power envelope of the speech signal to restore the temporal power envelope with this method. Then, power thresholding as a VAD decision was designed based on the restored temporal power envelope. A method of estimating the signal to noise ratio (SNR) was proposed to accurately estimate the SNR in the noise reduction stage. Experiments under both artificial and realistic noisy reverberant conditions were carried out to evaluate the performance of the proposed method of VAD and it was compared with conventional VAD methods. The results revealed that the proposed method significantly outperformed the conventional methods under artificial and realistic noisy reverberant conditions.     

基于遗传算法的声源三维定位方法

相位变换加权的可控响应功率（SRP-PHAT）算法是一种基于麦克风阵列的鲁棒声源定位方法,该算法在有混响和噪声的环境下仍有较高的定位精度.但该算法用网格法对整个声源空间进行搜索,逐点计算其目标函数,因而总的计算量非常大,不适用于实时定位系统.针对SRP-PHAT的特点,采用遗传算法进行搜索,使总的计算量大幅度降低.仿真结果表明在混响时间为300ms,信噪比为5dB的条件下,该算法仍可达到较高的定位精度.     

A fast search method of steered response power with small-aperture microphone array for sound source localization

The Steered Response Power （SRP） method works well for sound source localization in noisy and reverberant environment. However, the large computation complexity limits its practical application. In this paper, a fast SRP search method is proposed to reduce the computational complexity using small-aperture microphone array. The proposed method inspired by the SRP spatial spectrum includes two steps： first, the proposed method estimates the azimuth of the sound source roughly and determines whether the sound source is in far field or near field; then, different fine searching operations are performed according to the sound source being in far field or near field. Ex- periments both in simulation environments and real environments have been performed to compare the localization accuracy and computation complexity of the proposed method with those of the conven- tional SRP-PHAT algorithm. The results show that, the proposed method has a comparative accuracy with the conventional SRP algorithm, and achieves a reduction of 93.62% in computation complexity compared to the conventional SRP algorithm.     

Invariant‐Feature Based Object Tracking Using Discrete Dynamic Swarm Optimization

With the remarkable growth in rich media in recent years, people are increasingly exposed to visual information from the environment. Visual information continues to play a vital role in rich media because people's real interests lie in dynamic information. This paper proposes a novel discrete dynamic swarm optimization (DDSO) algorithm for video object tracking using invariant features. The proposed approach is designed to track objects more robustly than other traditional algorithms in terms of illumination changes, background noise, and occlusions. DDSO is integrated with a matching procedure to eliminate inappropriate feature points geographically. The proposed novel fitness function can aid in excluding the influence of some noisy mismatched feature points. The test results showed that our approach can overcome changes in illumination, background noise, and occlusions more effectively than other traditional methods, including color‐tracking and invariant feature‐tracking methods.     

基于频带间相关性的加权噪声功率谱估计

噪声功率谱估计是语音增强系统的一个重要组成部分。本文在加权噪声估计的基础上,考虑了带噪语音在相邻频带间的相关性,提出了一种新的噪声功率谱估计算法。该算法保留了加权噪声估计算法的优点,利用频域平滑及时域平滑后的带噪语音来求加权因子,能够更好地区分弱语音与噪声,尤其是对强语音后的弱语音与噪声区分更明显,从而具有更快的跟踪速度及更少的噪声过估计。客观实验和主观实验都证实了本文提出的算法的有效性。     

A New Post-filter Algorithm Combined with Two-step Adaptive Beamformer

The optimal microphone array, in the sense of minimum mean square errors (MMSE), includes two processing blocks: the minimum variance distortionless response (MVDR) beamformer and the single-channel Wiener filter, which acts as post-filter. In this paper, we propose a new post-filter algorithm based on assumptions that both the noise power attenuation factor (NPAF) and signal power attenuation factor (SPAF) are time invariant in the reverberant room. The algorithm recursively estimates both factors from available measurements and uses them in estimation of the post-filter parameters. Additionally, to overcome the problem of the poor performance of the MVDR beamformer in reverberant conditions, we propose the usage of the two-step (TS) MVDR algorithm. This algorithm improves the robustness of the beamformer and its ability to suppress the interferences using an estimate of the desired speaker transfer function. Although TS MVDR beamformer and proposed post-filter can work separately, or combined with other algorithms, the best performance is obtained when they work together. The performance of the proposed combination of new post-filter algorithm and TS MVDR beamformer is tested in a simulated reverberant room and compared with similar algorithms, which gave rather good results.     

基于反幂法和卡尔曼滤波的自适应语音去混响方法

噪声鲁棒的自适应语音信号去混响是现代语音信号处理的重要研究内容,其困难在于语音信号的非白性、非平稳性及房间的超长冲激响应特性。针对单输入多输出(SIMO)麦克风阵列系统获取的多路混响语音信号,提出了一种新的去混响算法。首先通过相关法时延估计对SIMO混响语音信号进行时延对齐;其次在保持SIMO系统输出信号间交叉关联关系(cross relation)基础上对混响语音信号进行预白化处理;最后把交叉关联关系、用于矩阵最小特征向量计算的反幂法与卡尔曼滤波解卷积方法有机结合,实现了SIMO混响语音信号的实时自适应去混响。仿真与实验研究表明,本方法对混响语音信号去混响效果明显,同时具有较好的抗噪声性能。      

Noisy Blind Signal-jamming Separation Algorithm Based on VBICA

Aiming at the blind signal-jamming separation (BSJS) in wireless communication environment, we propose a noisy BSJS based on Variational Bayesian Independent Component Analysis algorithm to separate the communication signal from jamming signals and noises. This algorithm takes the Kullback–Leibler divergence between the true post distributions of source signals and the approximate ones as objective function, models sources using mixture of Gaussians, and updates parameters of the model using variational-Bayesian learning method, so as to make the estimated approximate posterior distributions close to the true ones and recover source communication signals finally. The simulation results show that the proposed algorithm is effective for the BSJS in noisy environment.     

A new algorithm on multiple unknown source signals estimation and separation in a reverberant space

In many voice-related applications, the presence of echoes and overlapping speech signals can degrade the quality or intelligibility of a desired speech signal to be processed. It is, therefore, important to cancel the echoes and to separate overlapping speech signals from a mixture of these components, so that a specific function of the system, for instance, transmission, speech identification, or recognition, can be accomplished with better performance. However, in many cases we do not know the properties of the communication channel, and sometimes even the number of speech sources is unknown. In this paper, we propose to use a reference signal to determine the channel characteristics. When the estimated channel parameter matrices are obtained, a recurrence formula can then be used to separate various speech signals including their reverberant counterparts. As a finite impulse response (FIR) model is used to describe the observation model of the sources in the reverberant environment, it is not necessary for the processing speech signals to be uncorrelated. Because it involves only simple computation, our approach can be used in online applications. In this paper, we will investigate the validity of our algorithm and compare it with extended fourth-order blind identification (EFOBI). It is found that our method preserves both signal waveforms and their amplitudes even in a noisy environment, whereas EFOBI has not been able to achieve similar performance.     

自调整分层卡尔曼粒子滤波的快速目标跟踪

分层卡尔曼粒子滤波成功应用于目标跟踪,但其只对目标位置进行了优化,忽略了其他仿射参数,导致跟踪中的粒子数目仍然很大。为了实现复杂环境下的快速目标跟踪,提出一种带有自调整策略的分层卡尔曼粒子滤波方法。该方法将目标划分为线性和非线性状态空间,并通过少量粒子的迭代过程在非线性状态空间逐步搜索最优状态。其详细过程如下:首先,利用卡尔曼滤波预测目标位置,结合目标运动信息计算潜在目标区域;然后在该区域内生成一组随机粒子,通过在线姿态估计对粒子状态进行调整,并将观测结果与目标模板进行比较,修正粒子摄动的方向以逼近目标。把该方法应用于大机动目标的视频序列中,并与现有的跟踪方法进行了对比。结果表明,所提方法能够以少量粒子实现准确、稳定的目标跟踪,大大降低了跟踪算法的运算量,提高了跟踪效果。     

基于干扰观测的无线通信系统抗干扰功率控制算法

在快速变化的干扰环境下,无线通信系统传输可靠性会受到很大影响。为提升快速时变干扰环境下无线通信系统传输的可靠性,该文提出一种基于干扰观测的无线通信系统抗干扰功率控制算法。该算法首先将受到干扰影响的无线通信系统建模为广义稳定性控制系统,并采用干扰观测器生成系统状态受干扰影响的估计值。然后通过利用估计值来预测未来的跟踪误差和稳态的控制输入,优化系统的控制策略以实现对干扰环境的自适应调整。最后仿真结果表明,与传统方法相比,所提算法能够快速响应干扰变化,显著提高系统在快速时变恶意干扰下传输的可靠性,提高了系统对干扰环境的适应能力。     

利用频谱排序和筛选的突变噪声快速估计

针对最小值控制递归平均(Minima Controlled Recursive Averaging,MCRA)算法不能快速跟踪突变噪声的问题,提出了一种基于频谱排序和筛选的突变噪声快速估计方法。该方法在MCRA算法的基础上对带噪语音的功率谱进行排序,筛选出不含语音信号的频点来估计噪声的平均功率谱;当检测到噪声突变时,对当前的平滑参数和状态变量进行校正。仿真结果表明,该方法可以将突变噪声的跟踪时间缩短90%以上;用于语音降噪处理时,音质可以提升约0.4分。该方法具有一定的工程应用价值。     

Fast finite-time tracking control for a 3-DOF cable-driven parallel robot by adding a power integrator✰

Cable-driven parallel robots (CDPRs) usually suffer from kinematic and dynamic uncertainties, which makes it difficult for traditional trajectory tracking control algorithms to achieve high precision, fast response time, and robustness. In this study, we present a novel fast finite-time tracking control (FFTTC) algorithm which solves this problem to a large extent. Specifically, we firstly used a function of exponential errors with fractional power combined with API technique, to deal with the key difficulty of the convergence rate degradation which exists in traditional finite-time tracking control (TFTTC) when system states are far from the equilibrium point. Simultaneously, the API technique was used to avoid the problem of the explosion of complexity. To facilitate algorithm evaluation, the finite-time stability of the close system consisting of the proposed FFTTC algorithm and the CDPRs was proved mathematically and the settling time was estimated correspondingly. The trajectory tracking experiments were performed on a 3-DOF CDPR driven by 4 cables. Simulation and experimental results show that the proposed FFTTC algorithm can cope with external disturbances, variable load, and inaccurate model parameters. The comparison experiment indicates that the proposed FFTTC algorithm is superior to the model predictive control and TFTTC algorithms in precision, response speed and robustness.     

基于DSP的改进PSO光伏阵列MPPT控制应用

为了解决光伏（PV）系统在局部阴影条件下（PSC）的最大功率点跟踪问题,提出了一种基于改进粒子群算法（PSO）的快速最大功率点跟踪（MPPT）方法。与传统基于PSO的MPPT系统不同的是,采用了基于转换器电流动态行为的变量抽样时间策略（VSTS）,并且为了更快速的实现最大功率点跟踪,引入三个重要因数,即：粒子数、收敛速度以及抽样时间。采用DSP平台对提出系统进行了具体实现和性能评估,实验结果显示相比其他类似系统,在不同条件（包括PSC）下,提出算法均能够实现速度跟踪且精确度较高。     

Blind signals separation with genetic algorithm and particle swarm optimization based on mutual information

Blind source separation technique separates mixed signals blindly without any information on the mixing system. In this paper, we have used two evolutionary algorithms, genetic algorithm and particle swarm optimization for blind source separation. In these techniques a novel fitness function that is based on the mutual information and high order statistics is proposed. In order to evaluate and compare the performance of these methods, we have focused on separation of noisy and noiseless sources. Simulations results demonstrate that the proposed method for employing fitness function has rapid convergence, simplicity and a more favorable signal to noise ratio for separation tasks based on particle swarm optimization and continuous genetic algorithm than binary genetic algorithm. Also, particle swarm optimization enjoys shorter computation time than the other two algorithms for solving these optimization problems for multiple sources.     

真实环境中基于子帧分析的多说话人定位算法

针对真实环境中的多说话人定位问题,提出一种基于子帧分析的多声源定位算法。该算法将一帧语音信号分为8个子帧,利用每个子帧信号计算相位变换加权的可控响应功率函数,分别搜索其最大值得到声源位置的子帧估计。由于语音信号在时域具有稀疏性,这些估计值对应多个声源的位置。利用会聚聚类算法将子帧估计值分为若干类,然后用平均子帧可控响应功率函数对估计值进行评价,得到最终的声源位置估计。实验表明,在2~3个说话人的情况下,该算法比传统算法的定位性能有较大幅度提高。     

时变混合共轭梯度盲提取算法

针对传统独立分量分析(ICA)方法对时变信道跟踪能力较差的问题,提出了一种时变混合共轭梯度盲提取算法。该算法有效利用了各源信号的时序结构差异,仅利用其二阶统计量解决了具有不同功率谱密度的信号的分离,而无须估计信号的概率密度和计算高阶累积量,减少了运算的复杂度并可用于杂系信号混合的盲分离问题;同时,算法利用仅具有一个全局最优解的凸代价函数,采用计算简单并具有较好数值表现的自适应共轭梯度算法进行迭代,获得了更快的收敛速度和更好的稳定性能。仿真结果表明,该算法与传统ICA算法相比,具有对时变系统更好的跟踪能力。