麦克风阵列语音增强的研究

文章介绍了各种基本的麦克风阵列语音增强算法,对其消噪性能进行了系统地分析,并以实测数据进行了测试。并介绍了基于稳健波束形成器、近场超定向波束形成器、广义奇异值分解和传输函数广义旁瓣相消器等结构的麦克风阵列语音增强的基本原理,总结了各种算法的特点及其所适用的声学环境特性。     

多噪声环境下双微阵列语音增强算法

为提高双微阵列语音增强系统在多噪声环境下的消噪性能，提出一种适用于双微阵列的改进广义旁瓣抵消器语音增强算法。根据双微麦克风阵列的结构特点，首先，用基于噪声互功率谱估计的改进相干滤波算法消除距离较远麦克风之间产生的弱相关噪声；然后，利用广义旁瓣抵消算法消除距离较近麦克风之间产生的强相关噪声；最后，通过基于最小值控制递归平均的子带谱减法有针对性地消除不同频带上的残留噪声。仿真实验表明，在多噪声环境下所提算法较现有的双微阵列语音增强算法取得了更好的感知语音质量评价得分，一定程度上改善了双微阵列语音增强系统对复杂噪声的抑制效果。     

滤波器组自适应语音增强方法

双麦克风噪声抵消应用中,由于交叉串的存在,传统自适应算法降噪性能受到很大的影响。为了提高双麦克风算法降噪性能,使用两级自适应滤波系统消除交叉串扰问题。为提高自适应滤波器收敛性能,采用主从结构LMS算法自适应调节步长因子。同时为了适合窄带处理算法,将输入信号进行子带分析预处理,对每个子带独立进行抗交叉串绕自适应处理,将各子带增强信号合并得到增强语音信号。实验结果表明,该方消噪量大,语音损伤小,语音增强效果显著。     

一种广义旁瓣抵消器结构的语音增强改进算法

当广义旁瓣抵消器（Generalized sidelobe canceller,GSC）结构的语音增强算法对语音信号的入射方向角估计不准确时,阻塞矩阵（Blocking matrix,BM）不能完全阻塞目标语音,使得部分语音通过阻塞矩阵,在后期多输入抵消器（Multiple-input canceller,MC）模块中和参考信号相抵消,造成目标语音的损失。针对广义旁瓣抵消器因信号到达方向（Direction of arrival,DOA）估计误差而导致语音泄漏的问题,本文提出了一种麦克风阵列语音增强的优化算法,先对经过时延补偿的信号进行频谱调整,再利用MC模块输出与BM模块输出存在相关性的特点,对阻塞矩阵进行自适应调整,使方向估计参数更趋近于真实目标语音方向,以减少阻塞矩阵中目标语音的泄漏。仿真结果表明,该算法 可以有效减少阻塞矩阵中目标语音的泄漏、增强系统的鲁棒性以及提高语音增强效果。     

一种基于信噪比乘积更新的广义旁瓣抵消器

广义旁瓣抵消器由于实现结构简单在麦克风阵列算法中得到了广泛的应用.但是传统的广义旁瓣抵消器算法对导向矢量失配误差比较敏感,容易产生目标方向信号的自抵消现象.针对此问题,本文提出了一种改进算法,通过固定波束形成和自适应阻塞矩阵输出的信噪比乘积控制多输入抵消器的更新,降低目标方向信号的自抵消影响,提升语音质量.在8.5 m...     

民航卫星导航干扰信号多径抑制算法仿真

针对日益严重的卫星信号多径干扰问题,提出一种有效的民航卫星导航干扰信号多径抑制算法,在多径干扰卫星信号数据模型基础上,通过空间平滑技术采用损失天线阵列孔径方法实现多径卫星信号的解相干处理,得到不同阵列子阵数据协方差矩阵;再采用豪斯霍尔德变换的广义旁瓣相消技术,根据广义旁瓣相消器输出的多径和噪声功率以及不同阵列子阵数据协方差矩阵,计算直达信号最佳权重矢量,采用低复杂度自适应度算法迭代计算最佳权重矢量,获取最终旁瓣相消器多径抑制后的直达卫星信号,实现民航卫星导航干扰信号的多径抑制。仿真结果表明,所提算法在任意载噪比环境下的码相位跟踪误差均较小,对民航卫星导航干扰信号的多径抑制性能较强;且算法多径抑制耗时短、民航卫星导航的定位精度高。     

基于Kalman滤波的GSC改进语音增强算法

针对广义旁瓣相消器（Generalized sidelobe canceller,GSC）存在非相干噪声消除性能不佳的缺陷,提出了采用后置Kalman滤波器改进的GSC去噪算法。该算法通过归一化最小均方算法校正自适应噪声对消器,并将滤除方向性干扰噪声后的语音信号输出到Kalman滤波器中,对残余背景噪声进行迭代最小均方误差（Minimum mean square error,MMSE）估计,抑制非相干噪声与麦克风阵元所产生的热噪声。经过在不同信噪比条件下客观语音质量评估（Perceptual evaluation of speech quality,PESQ）及语谱图分析后证明,与传统的GSC以及后置谱减法的改进GSC相比,本算法在噪声消除上的表现更为优越,且增强后信号也更接近目标信号。     

基于麦克风阵列的语音增强算法研究

麦克风阵列信号处理技术的语音增强方法,能够充分利用语音信号的时空信息,其波束控制能力、抗干扰能力和信号增益均优于传统的方法。对于广义旁瓣抵消(GSC)的自适应滤波算法,在噪声相干的情况下具有很好的噪声抑制作用,但并不适用于噪声为非相干情形;反之,维纳滤波算法在噪声非相干的情况下对噪声有很好的抑制作用,而又不适用于噪声相干情形。为防止出现在高信噪比的情况下信号相消的现象,首先对GSC的阻塞矩阵进行了改进,其次对维纳滤波算法中信噪比取值的不确定性进行了改进,最后尝试将两种算法进行融合。仿真结果表明:融合算法在两种噪声情况下都具有较好的噪声抑制能力,在复杂噪声环境中具有更高的可靠性,因而更具实用价值。     

基于小波旁瓣相消器的轴承故障特征提取

为解决强背景噪声下声信号提取的轴承故障特征不显著问题,提出一种基于小波旁瓣相消器的故障特征提取方法。该方法利用小波滤波器组将含噪故障轴承声信号变换到小波域,进行小波域阵列广义旁瓣相消自适应波束形成,再通过小波滤波器组重构增强后的故障轴承信号,最后对重构增强后的信号进行包络解调并提取故障特征频率进行故障诊断。实验结果表明,该方法能够在强背景噪声下有效提取滚动轴承故障特征,并且相较于传统的延时求和波束形成器具有更好的降噪和故障特征增强效果。     

基于FPGA的双通道语音增强系统设计与实现

在数字助听器和小型语音设备的实际应用中,非平稳噪声干扰与自适应方法的收敛过程会造成语音性能下降。为了实际解决该问题,设计了一种新型的实时语音增强系统。该系统基于双通道一阶差分麦克风阵列,同时采用结构分时复用和高效汉宁窗分帧等方法,提高了性能并节约了硬件成本。该语音增强系统可获得3.5db左右的信噪比增益,同时克服了单通道增强系统和自适应方法的局限,并用Verilog语言在FPGA上设计实现该系统。从而在硬件层次上提高了小型语音设备的抗噪性能,为数字助听器或相关ASIC芯片的研制奠定了基础。     

基于麦克风阵列的语音增强技术及应用

用麦克风阵列进行语音处理的方法可以提高信噪比,解决环境噪声、回声和混响引起的语音识别性能降低的问题.介绍基于延迟-累加方法(传统波束法) 、自适应波束法及基于后置自适应滤波等结构的麦克风阵列语音增强的基本原理,总结了各种算法的特点.     

Blind Subband Beamforming With Time-Delay Constraints for Moving Source Speech Enhancement

A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.     

智能音箱中的一种快速回声消除算法

考虑到智能音箱中多采用麦克风阵列作为拾音装置,而单通道自适应滤波技术对声学回声消除具有失真性和复杂性,提出一种麦克风阵列快速回声消除算法。该算法首先用自适应滤波技术估计第一通道回声,然后估计阵列间的相对回声传递函数,把两者相乘得到其他通道回声;其次,把估计出的回声和噪声当作广义旁瓣抵消器（GSC）波束形成下支路的噪声参考信号,利用GSC波束形成算法去除回声和噪声。仿真结果表明,在中度混响、远距离、低回噪比且用音乐作为回声环境时,该算法具有良好的回声消除与噪声抑制性能,不仅运算量小,而且使目标语音信号具有较高的信源失真率和可懂度。     

基于麦克风小阵的多噪声环境语音增强算法

针对助听器等设备在非平稳或多种噪声并存环境下使用效果急剧下降的问题,提出一种基于小尺寸麦克风阵的相干滤波广义旁瓣抵消(CF-GSC)语音增强算法。该算法结合麦克风阵采集信号的特点,对各阵元间采集时表现为弱相关的海浪、风扇等近似白噪声,以及采集时表现为强相关的点源信号及其他竞争噪声,分别利用相干滤波和传统广义旁瓣抵消(GSC)结构对弱相关与强相关噪声的良好滤除效果,结合语音活动检测(VAD)在噪声段进行联合处理。仿真实验表明在多类噪声存在环境下,该算法能取得相对改进的通道间相干函数滤波算法及传统广义旁瓣抵消算法2 dB左右的增强效果提升,同时能获得良好的话音可懂度。     

A new adaptive filtering subband algorithm for two-channel acoustic noise reduction and speech enhancement

This paper addresses the problem of acoustic noise reduction and speech enhancement by adaptive filtering algorithms. Most speech enhancement methods and algorithms which use adaptive filtering structure are generally expressed in fullband form. One of these widespread structures is the Forward Blind Source Separation Structure (FBSS). This FBSS structure is often used to separate speech form noise and therefore enhance the speech signal at the processing output. In this paper, we propose a new subband implementation of this FBSS structure. In order to give more robustness to the proposed structure, we adapt then we apply to this subband structure a new combination of criteria based on the system mismatch and the smoothing filtering errors minimizations. The combination between this proposed subband structure with this optimal criteria allows to obtain a new two-channel subband forward (2CSF) algorithm that improves the convergence speed of the cross adaptive filters which are used to separate speech from noise. Objective tests under various environments are presented showing the good behavior of the proposed 2CSF algorithm.     

Hybrid probabilistic adaptation mode controller for generalized sidelobe cancellers applied to multi-microphone speech enhancement

In this paper, we propose a new adaptation mode controller (AMC) for a generalized sidelobe canceller (GSC)-based speech enhancement system. Here, a likelihood ratio for target speech presence was first estimated and then utilized to estimate both the local target speech presence probability (SPP) and global SPP. Next, the estimated SPPs were applied to the design of an AMC that controlled the parameters of adaptive filters for an adaptive blocking matrix (ABM) and noise canceller (NC). In particular, the combination of local and global SPPs was applied to the AMC in the ABM, whereas only global SPPs were used for the NC. Finally, a multiple-microphone speech enhancement system was constructed on the basis of a GSC having the proposed AMC. The performance of the speech enhancement system was subsequently evaluated in terms of the perceptual evaluation of speech quality (PESQ) and the cepstral distortion (CD) for car noise conditions. It was shown from this evaluation that a speech enhancement system using the proposed AMC method provided better performance than conventional AMC methods using power ratios between the target and non-target directional signals, the inter-channel normalized cross-correlation, and the local SPPs only.     

Car Speech Enhancement Using a Microphone Array

This paper proposes a speech enhancement approach to suppress the interference of car noise. A linear microphone array is adopted for far-talking speech acquisition and delay-and-sum beamforming noise reduction. We present an effective time delay estimator using the coherence function between the reference microphone and the beamformed speech. To further enhance the beamformed speech, we exploit an improved Wiener filter where the resulting noise correlation in microphone array is relatively small so that the performance of optimal Wiener filtering could be achieved. Also, due to the serious degradation in low frequency car speech, we develop a spectral weighting function to compensate the low frequency filtering. These two processing units serve as the post filters to attain the desirable enhancement performance. In the experiments on microphone array speech in presence of real and simulated car noises, we find that the proposed algorithm performs well. Performance is measured in terms of the signal-to-noise ratio and the word error rate. The combined delay-and-sum beamformer and two post filters obtain the best results compared to other methods.     

麦克风阵列语音增强技术及其应用

本文简要叙述了应用麦克风阵列进行语音增强的原理及方法。且由于麦克风阵列在实际语音处理时具有良好的拾取语音能力及噪声鲁棒性,本文将介绍该技术在车载系统环境、机器人语音识别、大型场所的记录会议、助听装置及声源定位等系统中的应用。     

Coherence based dual microphone speech enhancement technique using FPGA

This paper, presents a design and implementation of dual microphone coherence based speech enhancement technique using field programmable gate array (FPGA). In order to have a proper enhancement of dual microphone system, we require to estimate the time delay of arrival (TDOA) between the two microphone signals which is followed by the application of the proposed speech enhancement algorithm. We have used TDOA algorithm based on phase transform to minimize the effect of reverberation for localization of the sound sources. Coherence based technique has been used for speech enhancement process which requires no background noise estimation. In this way, we can achieve a high localization accuracy and also the capability of dealing with coherent noise. In the proposed system, TDOA and speech enhancement processes are executed concurrently exploiting the parallel logic blocks of FPGA, thus increasing the throughput of the system to a great extent. We have implemented our design on Spartan6 Lx45 FPGA device. The subjective evaluation of the proposed design with normal hearing listeners using comprehensibility listing test has been done and its performance has been compared to the existing state of the art research works. The objective evaluation of the proposed design also designates the significant melioration over the existing state of the art research works. The subjective and objective evaluation infer that our proposed hardware induce feasible solution for hearing aid and other hand-held devices.