声音定位机器人设计与实现

设计了基于麦克风阵列和时延估计算法的声音定位系统,硬件采用多通道同步模数转换器和数字信号处理器(DSP)实现;结合LabVIEW平台机器人,实现了通过远程控制机器人对声源进行实时定位跟踪。算法仿真和实际测试表明,该声音定位机器人定位跟踪能力良好。     

被动声探测在油罐机器人位置检测中的应用

为了定位油罐内作业机器人,利用被动声探测技术开发了机器人定位系统;设计了一个五元麦克风阵列拾取机器人发出的声信号,运用了改进的声源定位算法处理信号,进而完成油罐机器人的位置检测;给出了位置检测的算法流程,并搭建了实验平台;通过实验表明该方法可行并有实际应用价值。     

移动机器人三维空间听觉定位系统设计

研究移动机器人听觉系统优化问题,针对准确判断发声源位置,为提高机器人听觉系统在空间中的搜索能力,设计了一种基于声达时间差的移动机器人听觉定位系统.整个系统分为信号采集及预处理、捕捉声达时间点、定位计算和机器人控制四个部分.通过建立正四面体结构的麦克风阵列,感知三维空间中目标发出的声音信号,采用声电信号处理技术,实时捕捉声音到达时间点:对定位的数学模型测定目标的方位信息进行仿真.结果证明,听觉系统可实现机器人在三维空间中对声源目标的实时定位,定位精度较高.     

基于TMS320DM642麦克风阵列声源定位系统

麦克风声源定位是利用麦克风阵列拾取语音信号,并用数字信号处理技术对其进行分析和处理的声源定位技术.在麦克风阵列声源定位中,语音信号端点的拾取是重要的环节.语音端点检测是对接收到的信号利用端点检测算法分析,以确认麦克风阵列中语音信号到达的端点;并利用麦克风阵列中各麦克风接收到的语音信号的端点的先后,计算出麦克风阵列接收的...     

基于麦克风阵列的储罐内爬壁机器人定位技术

为提高储罐内爬壁机器人的智能水平及作业效率,研究设计了基于被动声定位技术的机器人定位系统.该系统利用麦克风阵列拾取机器人发出的声信号,运用改进的时延估计定位方法处理信号,从而定位爬壁机器人.介绍了系统涉及的语音信号处理方法,并利用卡尔曼滤波算法处理定位数据.实验表明15m内该系统的定位距离误差不超过12cm.     

针对头佩式麦克风阵列的声源定位算法研究

头佩式麦克风阵列在单兵便携反狙击声探测定位系统和机器人声定位系统中具有实际的应用价值。一般的声源定位方法是基于无遮挡的线性或非线性麦克风阵列。采用头佩式麦克风阵列,考虑到背向声源麦克风的低频声波由于头盔遮挡而发生的衍射作用,针对低频波段的声音信号进行定位算法的设计和研究。该算法利用低频声波的绕射路径计算时延,采用联合可控功率响应（SRP-PHAT）框架进行时延补偿搜索定位。实验表明,相比于普通的无遮挡定位算法,基于绕射路径的头佩式麦克风阵列定位方法通过综合利用背向声源的麦克风数据,明显地提高了定位的精度,这种精度的提升在选择1 kHz以内的信号频率窗口时达到最佳效果。     

基于麦克风阵列的声源被动定位系统设计

为了优化声源被动定位技术,以及降低功耗、提高精度、增强对人员不易到达的恶劣环境等区域的声源监测能力,设计了一种基于麦克风阵列的声源被动定位系统。以五元十字形为麦克风阵列模型,以FPGA为核心,采用VHDL编写逻辑控制程序对数据采集、读取过程进行控制,通过USB接口与上位机通信,主要设计了放大和接口电路模块。测试结果表明,系统定位距离准确,相对误差较小,满足系统设计要求。     

基于耳蜗核模型改进双耳时间差的声源定位

人耳听觉系统在噪声环境中能够准确定位感兴趣的声源,实现其定位的主要因素是双耳时间差,但是在噪声环境下利用双耳时间差方法进行定位的效果比较差。针对这一问题,提出一个基于耳蜗核模型的声源定位系统。利用耳蜗核模型模拟耳蜗核对听觉信息的处理机制,提取听觉神经纤维中对语声刺激同步的信息和发放率信息,从而实现对噪声的抑制,完成噪声环境下的声源定位。该系统在噪声环境下定位的误差为1.297°。实验结果证明改进之后的声源定位系统能在噪声环境下进行声源定位。     

基于TDOA和AOA算法的声源定位模型的研究

基于几何形状麦克风阵列的声源定位系统是近年来声源定位的研究热点之一,在众多领域应用广泛。本文在探讨已有的声源定位方法基础之上,对现有声源定位算法的优缺点进行整合,结合三维系统声源定位分布规律,以及TDOA和AOA算法构建仿真模型进行求解。     

危险气体泄漏源搜寻多机器人系统的设计与实现

危险气体泄漏源搜寻是仿生嗅觉技术的重要应用领域之一.为了提高气体泄露源定位的效率和准确性,设计并实现了一种基于无线传感器网络的气源目标搜寻多机器人系统.该系统由多个嗅觉机器人组成,每个机器人作为无线传感器网络节点实现信息交换,协同工作,实现危险气体泄漏源的定位.嗅觉机器人以DSP处理器(TMS320F28335)为控制核心,对MOS气体传感器和风速传感器的输出信号进行融合,设计了浓度梯度与风速信息相结合的单一气体泄漏源搜寻算法.当嗅觉机器人完成气源定位时将发出警报,其他机器人利用装配的麦克风阵列和声源定位算法实现对泄漏源的间接定位.最后,为了说明所设计的多机器人系统对气体泄露源定位的有效性和准确性,本文设计了针对单一泄露源的气源搜寻实验进行验证.     

A real-time super-resolution robot audition system that improves the robustness of simultaneous speech recognition

This study addresses a framework for a robot audition system, including sound source localization (SSL) and sound source separation (SSS), that can robustly recognize simultaneous speeches in a real environment. Because SSL estimates not only the location of speakers but also the number of speakers, such a robust framework is essential for simultaneous speech recognition. Moreover, improvement in the performance of SSS is crucial for simultaneous speech recognition because the robot has to recognize the individual source of speeches. For simultaneous speech recognition, current robot audition systems mainly require noise-robustness, high resolution, and real-time implementation. Multiple signal classification (MUSIC) based on standard Eigenvalue decomposition (SEVD) and Geometric-constrained high-order decorrelation-based source separation (GHDSS) are techniques utilizing microphone array processing, which are used for SSL and SSS, respectively. To enhance SSL robustness against noise while detecting simultaneous speeches, we improved SEVD-MUSIC by incorporating generalized Eigenvalue decomposition (GEVD). However, GEVD-based MUSIC (GEVD-MUSIC) and GHDSS mainly have two issues: (1) the resolution of pre-measured transfer functions (TFs) determines the resolution of SSL and SSS and (2) their computational cost is expensive for real-time processing. For the first issue, we propose a TF-interpolation method integrating time-domain-based and frequency-domain-based interpolation. The interpolation achieves super-resolution robot audition, which has a higher resolution than that of the pre-measured TFs. For the second issue, we propose two methods for SSL: MUSIC based on generalized singular value decomposition (GSVD-MUSIC) and hierarchical SSL (H-SSL). GSVD-MUSIC drastically reduces the computational cost while maintaining noise-robustness for localization. In addition, H-SSL reduces the computational cost by introducing a hierarchical search algorithm instead of using a greedy search for localization. These techniques are integrated into a robot audition system using a robot-embedded microphone array. The preliminary experiments for each technique showed the following: (1) The proposed interpolation achieved approximately 1-degree resolution although the TFs are only at 30-degree intervals in both SSL and SSS; (2) GSVD-MUSIC attained 46.4 and 40.6% of the computational cost compared to that of SEVD-MUSIC and GEVD-MUSIC, respectively; (3) H-SSL reduced 71.7% of the computational cost to localize a single speaker. Finally, the robot audition system, including super-resolution SSL and SSS, is applied to robustly recognize four sources of speech occurring simultaneously in a real environment. The proposed system showed considerable performance improvements of up to 7% for the average word correct rate during simultaneous speech recognition, especially when the TFs were at more than 30-degree intervals.     

基于机器人听觉的声源定位策略

针对机器人听觉定位,提出了五个传声器组成的阵列作为机器人的耳朵,其中四个传声器组成的平面阵确定声源空间位置,另外一个传声器辅助完成声源位于机器人前后方的判断,并在改进的时延算法上实现声源的空间定位。系统在室内环境下测试,实验结果证明在混响环境下机器人可以实现空间声源定位,该方法具有实时实现的有效性和应用性。     

Simultaneous Localization of a Mobile Robot and Multiple Sound Sources Using a Microphone Array

Sound source localization is an important function in robot audition. Most existing works perform sound source localization using static microphone arrays. This work proposes a framework that simultaneously localizes the mobile robot and multiple sound sources using a microphone array on the robot. First, an eigenstructure-based generalized cross-correlation method for estimating time delays between microphones under multi-source environments is described. Using the estimated time delays, a method to compute the farfield source directions as well as the speed of sound is proposed. In addition, the correctness of the sound speed estimate is utilized to eliminate spurious sources, which greatly enhances the robustness of sound source detection. The arrival angles of the detected sound sources are used as observations in a bearing-only simultaneous localization and mapping procedure. As the source signals are not persistent and there is no identification of the signal content, data association is unknown and it is solved using the FastSLAM algorithm. The experimental results demonstrate the effectiveness of the proposed method.     

Improved binaural sound localization and tracking for unknown time-varying number of speakers

A method based on the generalized cross-correlation (GCC) method weighted by the phase transform (PHAT) has been developed for binaural sound source localization (SSL) and tracking of multiple sound sources. Accurate binaural audition is important for applying inexpensive and widely applicable auditory capabilities to robots and systems. Conventional SSL based on the GCC-PHAT method is degraded by low resolution of the time difference of arrival estimation, by the interference created when the sound waves arrive at a microphone from two directions around the robot head, and by impaired performance when there are multiple speakers. The low-resolution problem is solved by using a maximum-likelihood-based SSL method in the frequency domain. The multipath interference problem is avoided by incorporating a new time delay factor into the GCC-PHAT method with assuming a spherical robot head. The performance when there are multiple speakers was improved by using a multisource speech tracking method consisting of voice activity detection (VAD) and K-means clustering. The standard K-means clustering algorithm was extended to enable tracking of an unknown time-varying number of speakers by adding two additional steps that increase the number of clusters automatically and eliminate clusters containing incorrect direction estimations. Experiments conducted on the SIG-2 humanoid robot show that this method outperforms the conventional SSL method; it reduces localization errors by 18.1° on average and by over 37° in the side directions. It also tracks multiple speakers in real time with tracking errors below 4.35°.     

Online Short-Term Multiple Sound Source Mapping for a Mobile Robot by Robust Motion Triangulation

The paper describes a two-dimensional (2-D) sound source mapping system for a mobile robot. The robot localizes the directions of sound sources while moving and estimates the positions of sound sources using triangulation from a short time period of directional localization results. Three key components are denoted. (i) Directional localization and separation method of different pressure sound sources by combining the Delay and Sum Beam Forming (DSBF) and the Frequency Band Selection (FBS) algorithms. (ii) The design of the microphone array by beam forming simulation to increase the resolution of the localization procedure and its robustness to ambient noise. (iii) Sound position estimation by using the RAndom SAmple Consensus (RANSAC) algorithm. Then we achieved 2-D multiple sound source mapping from time-limited data with high accuracy. Applying FBS as a binary filter after DSBF improves robustness for multiple sound source localization under robotic movement. In addition, a moving sound source separation method is shown by using segments of the DSBF enhanced signal derived from the localization process.     

基于神经网络的四元球面阵空间声源定向系统

针对移动机器人的目标声源定向问题,应用四元球面麦克风阵列建立了一套空间声源定向系统。为了解决通过阵列的时延值预测出目标声源方向的问题,提出了神经网络算法,并设计了双隐层BP神经网络。通过Matlab仿真验证了该网络可以实现近场和远场的声源定向,并在机器人本体上进行了实验测试,证明了该系统的实用性。     

SLAM-based Online Calibration for Asynchronous Microphone Array

Abstract

This paper addresses online calibration of an asynchronous microphone array. Although microphone array techniques are effective for sound localization and separation, these techniques have two issues; geometry information on a microphone array or time-consuming measurements of transfer functions between a microphone array and a sound source is necessary, and a fully synchronous multichannel analog-to-digital converter should be used. To solve these issues, we proposed an online framework for microphone array calibration by combining simultaneous localization and mapping (SLAM), and beamforming. SLAM simultaneously calibrates locations of microphones and a sound source, and clocks differences between microphones every time a microphone array observes a sound event. Beamforming works as a cost function to decide the convergence of calibration by localizing the sound using the transfer functions calculated from the estimated microphone locations and clock differences. We implemented a prototype system based on the proposed framework using extended Kalman filter-based SLAM and delay-and-sum beamforming. The experimental results showed that the proposed framework successfully calibrated an eight-channel asynchronous microphone array both in a simulated and a real environment even when system parameters such as variances are set to be 10 times larger than the optimal values. Furthermore, the error of sound localization with the calibrated microphone array was as small as the desired one, that is, the grid size for beamforming.     

Building ears for robots: Sound localization and separation

This paper describes our research on bio-mimetic robot audition. Among the many binaural and monaural sound localization cues in the human auditory system, the interaural time difference cue is selected as it can easily be obtained by omnidirectional microphones. We have used a three-microphone system to remove the anterior-posterior ambiguity which occurs in two-microphone (or ear) systems. The echo-avoidance model of the precedence effect is used to cope with the echoes and reverberations of real environments. We mimicked the cocktail party effect by perceptual grouping of continuous components according to the spatial information obtained by the sound localization method. A wheel-based mobile robot equipped with an auditory system was developed. The auditory system has two sound processing parts. One is a DSP-based realtime system; the other is an off-line system composed of remote computers. Experiments of localizing and separating multiple sound sources and robot navigation were conducted to demonstrate the system's ability and potential applications.