基于广义互相关时延估计算法的性能分析

广义互相关(GCC)算法是目前进行时延估计(TDE)的常用算法,文章首先对广义互相关时延估计算法中几种加权函数进行了分类论述,进而对部分加权函数进行仿真,最后通过综合比较揭示了它们各自的优缺点。     

基于信道频域模型及OFDM技术的超分辨率时延估计算法

就非自由空间电磁波测距这一课题,给出了信道的参数化频域模型,提出了互相关MUSIC超分辨率时延估计(TDE)算法,进一步结合先进的OFDM技术提出了基于信道频域响应及OFDM技术的互相关MUSIC算法,利用OFDM这一特殊的调制解调技术方便可靠的获得了信道的频域响应.对该算法的时延估计性能进行了仿真,结果表明该算法在[SNR]=0dB、[BW]=20 MHz条件下可实现精确的TDE,对应的测距误差不大于30 cm,充分体现了算法的超分辨率特性及抗噪性.     

基于麦克风阵列的GCC时延估计算法分析

准确的时延估计（Time Delay Estimation,TDE）是基于到达时间差（Time Difference of Arrival,TDOA）的声源定位技术的前提.在众多时延估计算法中,广义互相关（Generalized Cross Correlation,GCC）算法因其较低的运算复杂度和易于实现的特点得到了广泛的应用.针对不同的噪声情况,GCC时延估计算法利用不同的加权函数来抑制噪声干扰.本文在介绍麦克风阵列模型和GCC时延估计算法的基础上,针对GCC算法的弊端提出了一种改进算法,并在多种信噪比条件下,对部分加权函数的GCC时延估计算法进行了MATLAB仿真,通过比较其时延估计性能和声源定位精度,分析了这些加权函数各自的优劣性.     

基于容积粒子滤波的时延差定位估计算法

在基于粒子滤波的时延差定位估计方法中, 重要密度函数的选取将直接影响估计的性能, 为此, 提出了基 于容积粒子滤波的时延差估计(BCPF-TDE) 算法. 该算法利用最新的数据检测信息, 通过容积卡尔曼滤波(CKF) 获 取粒子滤波的重要性密度函数. 仿真实验表明, 在粒子数目相同的情况下, 基于容积粒子滤波的时延差估计(BCPF- TDE) 方法与基于扩展粒子滤波的时延差估计(BEPF-TDE) 方法相比, 定位估计误差只有后者的50% 左右, 而运行时 间相当.

     

声源定位中广义互相关时延估计算法的研究

基于时延估计(TDE)的声源定位算法是数字助听器中的核心算法之一,其估计精度会受到噪声和采样频率等因素的影响,导致了定位的不准确性。针对这一问题,结合相关峰精确插值算法(FICP),提出了一种基于二次相关改进的广义互相关时延估计算法。该算法通过二次相关,有效地降低噪声的干扰,再利用FICP,提高相关函数的分辨率。仿真实验表明,无论在低信噪比,还是在高信噪比环境下,改进算法的时延估计性能都有了明显改善。     

泄漏信号定位中的时延估计算法

时延估计是阵列信号处理中的一项关键技术,广泛用于如语音增强,说话人定位,其目的是要估计出同源信号到达不同传感器时,由于传输距离不同而引起的时间差。现有的算法主要包括相关、广义互相关方法、自适应最小均方方法等。这些算法因其抗噪性能不同,有着各自的应用场合。本文在一段公共的声波数据上利用各种算法进行时延估计,分析比较算法性能,并将相关方法应用于管道泄漏检测定位误差不超过5%。     

改进线性拟合的调制信号载波频率估计算法

为提高基于线性拟合的频率估计算法的抗噪性,根据调制信号的时域特征,在基于最小二乘法的基础上,研究了将随机抽样一致(RANSAC)算法用于载波频率估计。利用RANSAC算法进行直线拟合,来完成调制信号载波频率估计中的参数拟合。以具有单一载波的数字调制方式振幅键控(ASK)和移相键控(PSK)得到的调制信号为研究对象,在Matlab环境下进行仿真实验,结果表明：与基于最小二乘法的方法相比,所提出的载波频率估计方法的误差率明显降低,同时具有较好的抗噪性。     

基于多帧运动估计的帧率提升算法

结合最大后验概率(MAP)估计的准确性和分层块匹配算法的快速性,研究了一种多参考帧运动估计算法,并提出了一种基于多帧运动估计的帧率提升(FRUC)系统方案。实验结果表明,基于该算法的内插帧无论从客观指标(时间复杂度、信噪比PSNR)还是主观质量(视觉效果)均优于现有常用方法,且算法复杂度较低,便于硬件实现。     

基于非空时隙数的无线射频识别标签估算算法

针对无线射频识别(RFID)系统中现有标签估计算法估计时间长、误差大的问题,提出了一种基于非空时隙数的标签估算方法。首先,分析了动态帧时隙ALOHA(DFSA)算法的系统模型,指出标签估算的必要性;其次,对当前存在的一些标签估计算法进行了研究,列举其存在的不足;再次,通过在不同帧长条件下对非空时隙平均数与待识别标签数的关系进行研究,得出两者之间存在着的不依赖于帧长的归一化曲线并将其运用于标签估计。而且通过引入精度需求,运用概率分析理论和折半查找的方法来确定不同标签总数下的轮询次数K;最后,对所提标签估计算法进行仿真,从估算精度和估算时间两个方面与现有的标签估算算法作了性能对比分析。仿真结果表明,该算法最大估计误差仅为1%,在帧长为128、标签数为400的情况下,相比Adaptive Slotted ALOHA Protocol(ASAP)、Fast Zero Estimation(FZE)、最大后验概率(MAP)估计算法,其误差率分别减少了66.7%、78.3%和72.2%;此外在识别相同数目标签的情况下,所提算法耗费的估计时间也明显少于上述3种算法。由此可见,基于非空时隙数的标签估算算法具有较高的估算精度和估算效率,能够对RFID系统中的待识别标签进行快速准确的识别。     

5G NR RSRP测量算法研究

RSRP(Reference Signal Receiving Power)是代表无线信号强度的关键参数,也是物理层测量的重要指标之一。当前5G NR中RSRP的测量算法多是均值法和其衍生算法,通过对现有算法的研究提出一种基于子集的共轭相乘算法。根据均值算法和共轭相乘算法进行测量值估计,并对相应算法进行测量结果的比较,进而得到子集共轭相乘算法在不同信噪比情况下可以使用不同子集数来提高RSRP估计值。将提出的新算法结合5G系统的特点与均值算法和共轭相乘算法进行比较分析,得到三种算法各自的特点。     

Acoustic Beamforming for Speaker Diarization of Meetings

When performing speaker diarization on recordings from meetings, multiple microphones of different qualities are usually available and distributed around the meeting room. Although several approaches have been proposed in recent years to take advantage of multiple microphones, they are either too computationally expensive and not easily scalable or they cannot outperform the simpler case of using the best single microphone. In this paper, the use of classic acoustic beamforming techniques is proposed together with several novel algorithms to create a complete frontend for speaker diarization in the meeting room domain. New techniques we are presenting include blind reference-channel selection, two-step time delay of arrival (TDOA) Viterbi postprocessing, and a dynamic output signal weighting algorithm, together with using such TDOA values in the diarization to complement the acoustic information. Tests on speaker diarization show a 25% relative improvement on the test set compared to using a single most centrally located microphone. Additional experimental results show improvements using these techniques in a speech recognition task.     

Narrowband signal detection techniques in shallow ocean by acoustic vector sensor array

This paper presents the formulation and performance analysis of four techniques for detection of a narrowband acoustic source in a shallow range-independent ocean using an acoustic vector sensor (AVS) array. The array signal vector is not known due to the unknown location of the source. Hence all detectors are based on a generalized likelihood ratio test (GLRT) which involves estimation of the array signal vector. One non-parametric and three parametric (model-based) signal estimators are presented. It is shown that there is a strong correlation between the detector performance and the mean-square signal estimation error. Theoretical expressions for probability of false alarm and probability of detection are derived for all the detectors, and the theoretical predictions are compared with simulation results. It is shown that the detection performance of an AVS array with a certain number of sensors is equal to or slightly better than that of a conventional acoustic pressure sensor array with thrice as many sensors.     

Robust speaker's location detection in a vehicle environment using GMM models.

Abstract-Human-computer interaction (HCI) using speech communication is becoming increasingly important, especially in driving where safety is the primary concern. Knowing the speaker's location (i.e., speaker localization) not only improves the enhancement results of a corrupted signal, but also provides assistance to speaker identification. Since conventional speech localization algorithms suffer from the uncertainties of environmental complexity and noise, as well as from the microphone mismatch problem, they are frequently not robust in practice. Without a high reliability, the acceptance of speech-based HCI would never be realized. This work presents a novel speaker's location detection method and demonstrates high accuracy within a vehicle cabinet using a single linear microphone array. The proposed approach utilize Gaussian mixture models (GMM) to model the distributions of the phase differences among the microphones caused by the complex characteristic of room acoustic and microphone mismatch. The model can be applied both in near-field and far-field situations in a noisy environment. The individual Gaussian component of a GMM represents some general location-dependent but content and speaker-independent phase difference distributions. Moreover, the scheme performs well not only in nonline-of-sight cases, but also when the speakers are aligned toward the microphone array but at difference distances from it. This strong performance can be achieved by exploiting the fact that the phase difference distributions at different locations are distinguishable in the environment of a car. The experimental results also show that the proposed method outperforms the conventional multiple signal classification method (MUSIC) technique at various SNRs.     

Near field acoustic localization under unfavorable conditions using feedforward neural network for processing time difference of arrival

Using time difference of arrival (TDOA) is one of the two approaches that utilize time delay for acoustic source localization. Combining the obtained TDOAs together with geometrical relationships within acoustic components results in a system of hyperbolic equations. Solving these hyperbolic equations is not a trivial procedure especially in the case of a large number of microphones. The solution is additionally compounded by uncertainties of different backgrounds. The paper investigates the performance of neural networks in modelling a hyperbolic positioning problem using a feedforward neural network as a representative. For experimental purposes, more than 2000 sound files were recorded by 8 spatially disposed microphones, for as many arbitrarily chosen acoustic source positions. The samples were corrupted by high level correlated noise and reverberation. Using cross-correlation, with previous signal pre-processing, TDOAs were evaluated for every pair of microphones. On the basis of the obtained TDOAs and accurate sound source positions, the neural network was trained to perform sound source localization. The performance was examined using a large number of samples in terms of different acoustic sensors setups, network configurations and training parameters. The experiment provided useful guidelines for the practical implementation of feedforward neural networks in the near-field acoustic localization. The procedure does not require substantial knowledge of signal processing and that is why it is suitable for a broad range of users.     

Plane-Wave Decomposition of Acoustical Scenes Via Spherical and Cylindrical Microphone Arrays

Spherical and cylindrical microphone arrays offer a number of attractive properties such as direction-independent acoustic behavior and ability to reconstruct the sound field in the vicinity of the array. Beamforming and scene analysis for such arrays is typically done using sound field representation in terms of orthogonal basis functions (spherical/cylindrical harmonics). In this paper, an alternative sound field representation in terms of plane waves is described, and a method for estimating it directly from measurements at microphones is proposed. It is shown that representing a field as a collection of plane waves arriving from various directions simplifies source localization, beamforming, and spatial audio playback. A comparison of the new method with the well-known spherical harmonics based beamforming algorithm is done, and it is shown that both algorithms can be expressed in the same framework but with weights computed differently. It is also shown that the proposed method can be extended to cylindrical arrays. A number of features important for the design and operation of spherical microphone arrays in real applications are revealed. Results indicate that it is possible to reconstruct the sound scene up to order p with p2 microphones spherical array.     

Probabilistic Structure from Sound

Auditory perception is one of the most important functions for robotics applications. Microphone arrays are widely used for auditory perception in which the spatial structure of microphones is usually known. In practice, microphone array calibration can be tedious and other devices or means are required. The structure from sound (SFS) approach addresses the problem of simultaneously localizing a set of microphones and a set of acoustic events that provides a great flexibility to calibrate different setups of microphone arrays. However, the existing method does not take measurement uncertainty into account and does not provide uncertainty estimates of the SFS results. In this paper, we propose a probabilistic structure from sound (PSFS) approach using the unscented transform in which the uncertainties of the PSFS results are also available. In addition, a probabilistic sound source localization approach using the PSFS results is provided to improve sound source localization accuracy. The ample results of simulation and experiments using low-cost, off-the-shelf microphones demonstrate the feasibility and performance of the proposed PSFS approach.     

基于FastICA的高精度多径时延估计算法

在高精度无线定位中,针对现有超分辨率时延估计算法在低信噪比时性能较差的问题,提出了一种基于FastICA的多径时延估计算法。利用噪声子空间对白化矩阵进行修正,并采用参考信号约束初始分离矩阵,算法可快速收敛至所需的首达径分量,最后利用参考信号提取首达径时延。仿真结果表明,在信噪比较低的多径环境中,与已有超分辨时延估计算法相比能够有效提高时延估计精度。     

A Minimum Distortion Noise Reduction Algorithm With Multiple Microphones

The problem of noise reduction using multiple microphones has long been an active area of research. Over the past few decades, most efforts have been devoted to beamforming techniques, which aim at recovering the desired source signal from the outputs of an array of microphones. In order to work reasonably well in reverberant environments, this approach often requires such knowledge as the direction of arrival (DOA) or even the room impulse responses, which are difficult to acquire reliably in practice. In addition, beamforming has to compromise its noise reduction performance in order to achieve speech dereverberation at the same time. This paper presents a new multichannel algorithm for noise reduction, which formulates the problem as one of estimating the speech component observed at one microphone using the observations from all the available microphones. This new approach explicitly uses the idea of spatial-temporal prediction and achieves noise reduction in two steps. The first step is to determine a set of inter-sensor optimal spatial-temporal prediction transformations. These transformations are then exploited in the second step to form an optimal noise-reduction filter. In comparison with traditional beamforming techniques, this new method has many appealing properties: it does not require DOA information or any knowledge of either the reverberation condition or the channel impulse responses; the multiple microphones do not have to be arranged into a specific array geometry; it works the same for both the far-field and near-field cases; and, most importantly, it can produce very good and robust noise reduction with minimum speech distortion in practical environments. Furthermore, with this new approach, it is possible to apply postprocessing filtering for additional noise reduction when a specified level of speech distortion is allowed.