Experiments on authenticity and plausibility of binaural reproduction via headphones employing different recording methods

Major criteria for a successful binaural reproduction are not only a suitable localization performance, but also the authenticity and plausibility of the presented scene. It is therefore interesting to examine whether the binaural reproduction can be perceptually distinguished from a real source. The aim of the presented investigation is to compare the quality of the binaural reproduction via headphones with two different microphone setups (miniature microphone in Open-Dome and ear plug) for individual head-related-transfer-function (HRTF) and headphone-transfer-function (HpTF) measurements. Listening tests with a total of 80 subjects were carried out focusing on plausibility and authenticity. In the examination of plausibility detection rates showed that subjects were not able to match the reproduced pink noise to its reproduction system (real source vs. binaural reproduction via headphones). The authenticity of the static binaural reproduction was highly dependent on the stimulus. Pink noise could often be distinguished due to coloration in higher frequencies and small differences in location. A difference between microphone setups could not be found in neither of the listening tests.     

微型数字传声器技术与发展

介绍了微型驻极体(ECM)与微机电(MEMS)数字传声器的技术原理及在消费数码领域的应用,综述了微型数字传声器技术的发展过程与市场分析。     

非均匀傅里叶变换频率不变波束形成方法

频率不变波束形成是宽带阵列信号处理研究的重要内容之一。本文以麦克风阵列语音处理为研究对象,针对FIB低频处波束形成一致性不理想的问题,提出了一种基于非均匀傅里叶变换的频率不变波束形成方法。该方法采用非均匀采样来增加低频处的采样点数,构建符合非均匀傅里叶逆变换的范德蒙矩阵,再用优化对称窗函数对变换结果进行截取,得到空时滤波器。该方法不仅实现了低频段与期望波束响应近似的波束图,且降低了阵元数目。计算机仿真实验结果验证了该方法的有效性。      

Robust tracking of moving sound source using multiple model Kalman filter

In this paper a novel method for tracking an active speaker in a noisy and reverberant environment by means of a spatially distributed microphone array is presented. Firstly, a sound source localization algorithm based on time delays of arrival (TDOA) in microphone pairs provides observed position estimates. Then these remarkably noisy estimates are filtered by a multiple model Kalman filter (MMKF) in order to obtain a smoothed trajectory of the speaker’s movement. Compared with the traditional Kalman filter (KF), simulated results prove the MMKF is more robust and effective in noisy environments.     

The implementation of low-cost urban acoustic monitoring devices

The urban sound environment of New York City (NYC) can be, amongst other things: loud, intrusive, exciting and dynamic. As indicated by the large majority of noise complaints registered with the NYC 311 information/complaints line, the urban sound environment has a profound effect on the quality of life of the city’s inhabitants. To monitor and ultimately understand these sonic environments, a process of long-term acoustic measurement and analysis is required. The traditional method of environmental acoustic monitoring utilizes short term measurement periods using expensive equipment, setup and operated by experienced and costly personnel. In this paper a different approach is proposed to this application which implements a smart, low-cost, static, acoustic sensing device based around consumer hardware. These devices can be deployed in numerous and varied urban locations for long periods of time, allowing for the collection of longitudinal urban acoustic data. The varied environmental conditions of urban settings make for a challenge in gathering calibrated sound pressure level data for prospective stakeholders. This paper details the sensors’ design, development and potential future applications, with a focus on the calibration of the devices’ Microelectromechanical systems (MEMS) microphone in order to generate reliable decibel levels at the type/class 2 level.     

Effects of Microphone Type on Acoustic Measures of Voice

Acoustic measures provide an objective means to describe pathological voices and are a routine component of the clinical voice examination. Because the voice sample is obtained using a microphone, microphone characteristics have the potential to influence the values of parameters obtained from a voice sample. This project examined how the choice of microphone affects key voice parameters and investigated how one might compensate for such microphone effects through filtering or by including additional parameters in the decision process. A database of 53 normal voice samples and 100 pathological voice samples was used in four experiments conducted in an anechoic chamber using four different microphones. One omnidirectional microphone and three cardioid microphones were used in these experiments. The original voice samples were presented to each microphone through a speaker located in an anechoic chamber, and the output of each microphone sampled to computer disk. Each microphone modified the frequency spectrum of the voice signal; this, in turn, affected the values of the voice parameters obtained. These microphone effects reduced the accuracy with which acoustic measures of voice could be used to discriminate pathological from normal voices. Discrimination performance improved when the microphone output was filtered to compensate for microphone frequency response. Performance also improved when spectral moment coefficient parameters were added to the vocal function parameters already in use.     

基于传声器阵列过顶测量结果的飞机起落架噪声研究

应用由111个传声器组成的平面传声器阵列对当前流行的民用客机进场着陆过程中的机体噪声源进行了实验测量,本对七架窄体客机和七架宽体客机的起落架噪声进行了分析,得到了起落架噪声的频谱特性、指向特性和声级变化。研究发现,起落架噪声的频谱是由宽频随机噪声与一些较为明显的单噪声源组成,起落架噪声的指向性类似于一个水平放置的偶极子。不同飞机起落架噪声的声级相差较大,这说明可以通过重新结构设计降低起落架噪声。     

Array signal processing with two outputs preserving binaural information

A new type of array signal processing combined with a weighted least squares algorithm to enable two-channel output with binaural information is proposed in this paper. This algorithm may be effective for use in a binaural hearing aid because the interaural relationship can be preserved after array signal processing. Retaining spatial information on specified directions while sufficiently suppressing unnecessary ambient noise coming from directions other than those of target sounds is required for this type of algorithm. In order to satisfy these two simultaneous requirements, the proposed algorithm was derived from a constraint algorithm by employing the weighted least squares algorithm. Performance in directivity patterns as well as interaural time difference (ITD) and interaural level difference (ILD) were evaluated. Computer simulations showed that this algorithm yields robust performance in various conditions compared with array signal processing based on a constraint algorithm.     

Design of robust nearfield wideband beamformers with optimum subband constraints

Design of nearfield wideband beamformers robust against microphone mismatches is of great interest in practical applications. The state-of-the-art design approach is based on the fullband processing. In this paper, the over-constraint problem suffered by the fullband design approach is studied, which typically leads to the undesired signal distortion with low-pass filtered characteristics. To combat the over-constraint problem, a new design approach for robust nearfield wideband beamformers with optimum subband constraints is proposed. The performance of the proposed design approach is evaluated and compared with the fullband design counterpart via design examples.