改进的同步迭代算法在光声血管成像中的应用

光声成像结合了光学成像和声学成像的优点，是一种高分辨率，高对比度的无损伤医学成像技术.一种改进的同步迭代算法应用于光声图像重建.仿真和模拟结果表明，与传统的代数迭代算法相比，在90°，135°，180°的有限场光声成像中，此算法对测量误差的校正和迭代次数的收敛上具有较大的优势，图像重建的速度和成像质量都有了明显的提高.实验中，一种圆形扫描结构的光声成像装置，用于180°的有限场扫描，利用改进的同步迭代算法，重建出了高对比度和高分辨率（60μm）的鸡胚胎光声血管图像.实验证明，这种算法的应用，大幅度减少了数据采集时间，为光声成像技术运用于实时监测血流灌注和肿瘤光动力治疗的血管损伤效应提供了潜在的应用价值. 关键词： 光声成像 有限角度 代数迭代算法 光声血管成像     

基于维纳滤波反卷积的光声成像

为了提高光声成像(PAT)的对比度和分辨率,需对组织样品的光声(PA)信号进行基于探头脉冲响应的滤波反卷积以恢复其频谱特性.对宽带光声信号而言,由于带通滤波的截止频率由人为确定,噪声不能得到有效抑制,很难获得稳定的反卷积结果.针对此问题,提出了基于维纳滤波反卷积的光声成像方法,利用点光声源获得超声探头的脉冲响应.利用维纳滤波抑制反卷积过程中噪声的影响,滤波器参数由离散小波变换(DWT)动态估计,样品光声图像由时域后向投影算法重建.数值模拟与成像实验均表明该方法有效地抑制了噪声对反卷积的影响,提高了光声成像的对比度和分辨率.     

光声光谱检测装置中光声池的数值计算及优化

利用光声光谱技术进行痕量气体的检测具有独特的优势,光声池是系统装置中最为重要的核心部件,它决定着整机性能的优劣.以一圆柱形共振型光声池为研究对象,基于声学与吸收光谱学的基本理论,建立了光声池声场激发的数学模型;利用数值模拟方法对光声池空腔结构进行了声学模态仿真,获得了前8阶声学模态值以及声压可视化振型;在考虑热黏性声学损耗的作用下,对光声池进行了热-声耦合多物理场仿真计算;将仿真结果与解析计算和实验结果进行对比,明确了利用数值模拟方法来计算光声池有关指标的可靠性与可行性;针对光声池的优化问题,提出了一种将响应面代理模型与遗传算法相结合的优化算法,在将原光声池中的谐振腔两端形貌更改为喇叭口形的情况下,通过优化算法获得了以光声池品质因数Q及池常数C_(cell)为最大值寻优的Pareto最优解集;选取一组解进行考察,结果表明,代理模型预测值与数值模拟值指标最大误差仅为1.3%,优化后的新型光声池Q较之前增长了48.9%, C_(cell)增长了34.4%.研究方法可为光声光谱中光声池的优化设计提供参考借鉴.     

小阻尼界面房间声传输函数和声脉冲响应函数的有限元素法

有限元法可用于以声波动方程为基础通过数值计算求解室内声场,适用于分析界面阻抗非均匀分布和复杂形状房间内声场的低频特性。本文首先介绍了小阻尼界面条件下室内声场简正方式、声衰变系数、混响时间的ＦＥＭ计算方法。在此基础上导出了房间内两点之间声传输函数和声脉冲响应函数的ＦＥＭ计算模型,并以矩形房间为例详细讨论了有关细节。本文所讨论的计算模型可以映房间内不同的声源点,接收点位置上的声压频谱特性和脉冲响应的时     

利用声辐射模态重构任意目标的散射声场

水下目标散射声场的重构可以作为水下目标散射特性的研究基础。本文主要利用声辐射模态对水下目标进行散射声场重构研究。首先,在借助声传递矩阵给出的任意结构声辐射模态的流体域求解方法基础上,通过理论证明了目标的散射声压与声辐射模态具有函数关系。其次,借助声场分布模态的概念,同时考虑到声场分布模态病态及声压测量易受噪声污染,提出基于声辐射模态的正则化散射声场重构算法。仿真结果表明,波数越低,重构所需声辐射模态阶数越少,在较高波数时仅需总模态数的大约20%即可对声场进行重构。与基于边界元的声场重构算法相比,计算量减小了至少80%,且克服了赫姆霍兹积分方程最小二乘法仅对球壳结构的重构效果较好而不适用于长条形结构重构的缺陷。     

水滴声台阶声景观软件模拟实验*

国内外发现了一种能够将脚步声或拍手声转变成类似水滴声的水滴声台阶。为科学保护和实现主动设计这种水滴声台阶声景观,该文基于声学仿真软件对天子冢和天元山水滴声台阶声场特征进行模拟,讨论了模拟的精确性和水滴声台阶形成的复杂性。结果表明:当忽略台阶物理属性影响时,可以模拟出台阶几何形式对声音频谱有一定的筛选作用,模拟出的台阶上的声场特征与实地测试结果基本一致,但是无法准确模拟台阶上"水滴声"频谱的主导频率大小。最后提出水滴声台阶保护和设计建议。     

基于声矢量传感器的分布式定位系统在水下宽带声源定位中的应用

提出了基于声矢量传感器的分布式浮标网络定位系统,研究了不同应用背景下单个声矢量传感器的测向算法,推导了目标DOA估计的Cramer-Rao界,给出了分层海水介质中多个声矢量传感器的几何定位算法。数值仿真结果表明:(1)系统的定位性能强烈依赖于接收信噪比;(2)该系统适用于单一强声源的定位。     

一种声发射信号方位角自适应监测算法*

四阶累积量可用于声发射信号处理,结合盲波束形成算法可对声发射信号的方位角进行估计。由于四阶累积量的计算量较大,采用迭代计算的思想,提出一种基于四阶累积量的自适应声发射信号方位角监测算法。与改进前的算法相比,自适应算法的计算复杂度和所需存贮空间均显著降低,可用于声发射信号的方位估计和监测。计算机仿真及实测数据分析均验证了算法的有效性。     

光声信号的声透镜层析成像研究

提出了一种用声透镜实现光声层析成像的新模式。从理论上计算出了声透镜的响应，测出了已知声场中标准物像面处的声场分布。考虑到圆形活塞振源的指向性，对代表物成像进行了理论修正，并与实验结果做了对比分析。研究表明，利用声透镜可以实现光声层析成像，并经图像重构得到了生物组织中异物的光声图像，横向、纵向分辨力较高。     

利用维纳滤波改善声透镜光声成像系统的分辨率

为了克服衍射效应对光声成像系统分辨率的限制,需要采用逆卷积方法进行图像反演.从理论上分析了声透镜成像原理,模拟仿真了声透镜的点扩展函数对声透镜成像系统分辨率的影响和维纳滤波解卷积方法复原光声成像的过程,并利用自搭建的声透镜光声成像系统进行了深入的实验研究,得到了物平面上相距4 mm和3 mm的两个黑胶带点的直接成像光声...     

Perceptual validation of virtual room acoustics: Sound localisation and speech understanding

The reliability of algorithms for room acoustic simulations has often been confirmed on the basis of the verification of predicted room acoustical parameters. This paper presents a complementary perceptual validation procedure consisting of two experiments, respectively dealing with speech intelligibility, and with sound source front–back localisation.The evaluated simulation algorithm, implemented in software ODEON®, is a hybrid method that is based on an image source algorithm for the prediction of early sound reflection and on ray-tracing for the later part, using a stochastic scattering process with secondary sources. The binaural room impulse response (BRIR) is calculated from a simulated room impulse response where information about the arriving time, intensity and spatial direction of each sound reflection is collected and convolved with a measured Head Related Transfer Function (HRTF). The listening stimuli for the speech intelligibility and localisation tests are auralised convolutions of anechoic sound samples with measured and simulated BRIRs.Perception tests were performed with human subjects in two acoustical environments, i.e. an anechoic and reverberant room, by presenting the stimuli to subjects in a natural way, and via headphones by using two non-individualized HRTFs (artificial head and hearing aids placed on the ears of the artificial head) of both a simulated and a real room.Very good correspondence is found between the results obtained with simulated and measured BRIRs, both for speech intelligibility in the presence of noise and for sound source localisation tests. In the anechoic room an increase in speech intelligibility is observed when noise and signal are presented from sources located at different angles. This improvement is not so evident in the reverberant room, with the sound sources at 1-m distance from the listener. Interestingly, the performance of people for front–back localisation is better in the reverberant room than in the anechoic room.The correlation between people’s ability for sound source localisation on one hand, and their ability for recognition of binaurally received speech in reverberation on the other hand, is found to be weak.     

Locating animals from their sounds and tomography of the atmosphere: experimental demonstration.

Calling animals are located using widely distributed receivers, and the sounds from the animals are used to map the sound speed and wind fields by means of tomography. In particular, two Red-Winged Blackbirds Agelaius phoeniceus are correctly located within a meter using recordings from five receivers spread over a 20 by 30 m region. The demonstration hinges on two new developments. First, a new algorithm for blindly estimating the impulse response of the channel is shown capable of estimating the differences in the time of first arrivals at two receivers. Since it is known that the first arrivals travel along nearly straight paths, the difference in time constrains the animal's location to a hyperboloid, and the animal is located by intersecting hyperboloids from many pairs of receivers. Second, in order to accurately find the intersection point and map the sound speed and wind fields using tomography, a nonlinear equation is solved. The new algorithm for blindly estimating the impulse response of a channel offers a new way for locating sounds and making tomographic maps of the environment without any requirement for a model for the propagation of sound such as is needed for focalization and matched field processing.     

基于中值滤波和提升小波分析的图像去噪方法研究

针对现有算法大多对单一高斯噪声或脉冲噪声进行图像滤波的问题,在对二维图像平滑去噪的过程中,采用基于中值滤波和提升小波变换相结合的图像去噪方法。在中值滤波基础上,构造基于脉冲检测的中值滤波器,找出混合噪声中脉冲噪声并进行滤波;与此同时,对原始小波进行提升,构造提升小波,然后采用提升小波阈值去噪方法抑制高斯噪声。实验结果表明:采用本文方法,混合噪声得到有效抑制,去噪效果好。     

Feasibility of subjective speech intelligibility assessment based on auralization

Subjective speech intelligibility can be assessed by speech recorded in an anechoic chamber and then convolved with room impulse responses that can be created by acoustic simulation. The speech intelligibility (SI) assessment based on auralization was validated in three rooms. The articulation scores obtained from simulated sound field were compared with the ones from measured sound field and from direct listening in rooms. Results show that the speech intelligibility prediction based on auralization technique with simulated binaural room impulse responses (BRIRs) is in agreement with reality and results from measured BRIRs. When this technique is used with simulated and measured monaural room impulse responses (MRIRs), the predicted results underestimate the reality. It has been shown that auralization technique with simulated BRIRs is capable of assessing subjective speech intelligibility of listening positions in the room.     

A system for simulating room acoustical environments for one’s own voice

The real-time simulation of room acoustical environments for one’s own voice using generic software has been difficult until very recently due to the computational load involved: requiring real-time convolution of a person’s voice with a potentially large number of long room impulse responses. This paper describes a software-based solution that accomplishes real-time convolution with head-tracking to simulate the effect of room acoustical environments on the sound of one’s own voice, using binaural technology. Actual rooms are characterized by measuring the room impulse response from the mouth to ears of the same head (oral binaural room impulse response, OBRIR). By repeating this process at 2° yaw increments for a given head position, the rooms are binaurally scanned around a given position to obtain a collection of OBRIRs, which is then used by the software-based simulation system. In the simulated rooms, a person equipped with a near-mouth microphone and near-ear loudspeakers can speak or sing and hear their voice, as it would sound in the recorded rooms, while physically being in an anechoic room. By continually updating the person’s head orientation using head-tracking, the corresponding OBRIR is chosen for convolution with their voice. The system described in this paper achieves the low latency that is required to simulate nearby reflections, and it can perform convolution with long room impulse responses.     

Matching pursuit of an adaptive impulse dictionary for bearing fault diagnosis

The sparse decomposition based on matching pursuit is an adaptive sparse expression of the signals. An adaptive matching pursuit algorithm that uses an impulse dictionary is introduced in this article for rolling bearing vibration signal processing and fault diagnosis. First, a new dictionary model is established according to the characteristics and mechanism of rolling bearing faults. The new model incorporates the rotational speed of the bearing, the dimensions of the bearing and the bearing fault status, among other parameters. The model can simulate the impulse experienced by the bearing at different bearing fault levels. A simulation experiment suggests that a new impulse dictionary used in a matching pursuit algorithm combined with a genetic algorithm has a more accurate effect on bearing fault diagnosis than using a traditional impulse dictionary. However, those two methods have some weak points, namely, poor stability, rapidity and controllability. Each key parameter in the dictionary model and its influence on the analysis results are systematically studied, and the impulse location is determined as the primary model parameter. The adaptive impulse dictionary is established by changing characteristic parameters progressively. The dictionary built by this method has a lower redundancy and a higher relevance between each dictionary atom and the analyzed vibration signal. The matching pursuit algorithm of an adaptive impulse dictionary is adopted to analyze the simulated signals. The results indicate that the characteristic fault components could be accurately extracted from the noisy simulation fault signals by this algorithm, and the result exhibited a higher efficiency in addition to an improved stability, rapidity and controllability when compared with a matching pursuit approach that was based on a genetic algorithm. We experimentally analyze the early-stage fault signals and composite fault signals of the bearing. The results further demonstrate the effectiveness and superiority of the matching pursuit algorithm that uses the adaptive impulse dictionary. Finally, this algorithm is applied to the analysis of engineering data, and good results are achieved.     

Development of a hybrid computer model for simulating the complicated virtual sound field in enclosures

A hybrid computer model, SOFIS, has been developed for the simulation of an enclosed virtual sound field in an arbitrary shaped enclosure. It can be used to calculate the impulse response and acoustical parameters of different positions in a virtual enclosure. This paper describes the way in which SOFIS models the sound source, the receiver and the sound propagation throughout the enclosed space with curved surfaces or barriers. A phase tracing method and the calculation of acoustic indexes are also discussed in this paper.     

基于单片机的声脉冲信号发生器设计

声脉冲法是近代现场隔声测试和吸声测试技术的主要方法之一,但声脉冲信号的产生是阻碍其应用的一个难点。本文以AT89C52单片机和DAC0832模∕数芯片为核心,设计一种声脉冲信号发生器,具有结构简单,成本低廉,使用方便的特点。经过测试,给出了该声脉冲信号发生器所产生声脉冲信号的时频域波形图。     

A simple impulse sound source for measurements in room acoustics

This paper deals with the construction and characteristics of a specially designed wooden clapper intended to operate as an impulse sound source for measurements in rooms. It is to be used as an alternative impulse sound source for experimental estimations of impulse responses of rooms located at remote and specific places where none of the standard sound sources - an omnidirectional loudspeaker system, etc. - can be used. The paper describes the spectral characteristics and directivity of the clapper impulse. Its features are compared with other impulse sources.     

Research on intensity measurement in room impulse field

A new system of sound intensity measurement for impulse field in the room was proposed. This measurement system consists of a repeatable inspiriting sound source and a microphone fixed on a slowly rotating platform, which is equivalent to a circle microphone array composed of many perfectly matched microphones. The test principle was presented and typical application was described. Based upon this system the sound intensity measurement for impulse field in the room was realized. Therefore, not only time but also spatial information of room impulse response can be obtained.