A new definition of boundary point between early reflections and late reverberation in room impulse responses

The early reflections in the room impulse response are usually defined as those observed within the initial 80 ms after the arrival of the direct sound, after which time the sound field is called reverberant. This number was chosen from measurements of other functions in a limited number of halls. In order to give an objective foundation to this time separation and to establish a physical indicator for it, a new method is proposed that defines a "transition time t(L)," which is the time at which the energy correlation between the direct plus initial sound and the subsequent decaying sound first achieves a specified low value. For various halls this number is shown and its relevance as a new parameter is discussed.     

Reliability of estimating the room volume from a single room impulse response

The methods investigated for the room volume estimation are based on geometrical acoustics, eigenmode, and diffuse field models and no data other than the room impulse response are available. The measurements include several receiver positions in a total of 12 rooms of vastly different sizes and acoustic characteristics. The limitations in identifying the pivotal specular reflections of the geometrical acoustics model in measured room impulse responses are examined both theoretically and experimentally. The eigenmode method uses the theoretical expression for the Schroeder frequency and the difficulty of accurately estimating this frequency from the varying statistics of the room transfer function is highlighted. Reliable results are only obtained with the diffuse field model and a part of the observed variance in the experimental results is explained by theoretical expressions for the standard deviation of the reverberant sound pressure and the reverberation time. The limitations due to source and receiver directivity are discussed and a simple volume estimation method based on an approximate relationship with the reverberation time is also presented.     

超声探头的响应特性及暂态回波模型

建立检测系统的数学模型,可以更好地理解超声检测的物理本质。分析了超声波从产生、介质中传播、缺陷耦合以及接收的全过程,将缺陷回波表示为探头响应函数与缺陷响应的时域卷积。利用空间脉冲响应和基尔霍夫近似建立了超声波与平面型缺陷的耦合模型,用大平面试块底面回波和大平面响应进行反卷积求得了探头的响应函数,并详细分析了探头在不同偏置位置时不同大小缺陷响应的特点,发现缺陷回波由直达回波和边缘回波组成,直达回波和边缘回波极性相反,且直达回波的幅值远远大于边缘回波。      

A spatial impulse response based method for determining effective geometrical parameters for spherically focused transducers

This paper proposes a novel method for determining effective geometrical parameters (GPs) of spherically focused transducer. Theoretical basis of the method is the spatial impulse response method (SIRM) that is a powerful tool to calculate transient fields from a piston-like transducer. Experimentally, the method is implemented by measuring arrival times of the pulse signals (either transmission or pulse echo signals) and using them to determine the time intervals between the direct and edge wave pulses. Since the time intervals for a given measurement position depend on the transducer's GPs, then the effective GPs can be obtained. The measurements are carried out in the near-field so that the time intervals can be resolved. The proposed method has been applied to a spherically focused transducer operating in pulse echo mode. The results have shown that the effective GPs used in the SIRM give a closer agreement between the theory and the measurements than the nominal GPs. The method is best suited for characterizing broadband transducers with short pulse excitations. With certain modification it can be also applied to narrowband transducers. The method can be also applied to transducers of other shapes, e.g., planar circle, and planar or curved rectangle provided the piston approximation is appropriate.     

Reconstruction of a complex-shaped defect from the known arrival time of the reflected ultrasonic wave

The problem of reconstructing the boundary of an arbitrarily shaped defect formed inside an elastic body from the measured time of arrival of the reflected ultrasonic wave in the echo method is considered. The characteristic size of the defect is assumed to be greater than the wavelength, and the defect is irradiated from the far-field zone, which means that the incident wave can be considered as plane. An algorithm is developed for reconstructing the convex envelope of a nonconvex defect from the arrival times of echo signals measured at different angles with the use of circular scanning.     

High-resolution interference acoustic location

A method of forward-scattering location on the basis of resolving the arrival times of direct and scattered signals is proposed. The method is specific in that it allows one to use a long sinusoidal probing signal. The required time resolution of the signals produced by moving targets is achieved by processing the spectrum of the received echo signals so as to broaden its effective bandwidth. The method is tested in both numerical and full-scale acoustic experiments. For the first time, the forward-scattering location is implemented with the use of long probing pulses in a narrow frequency band.     

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.     

Subjective and objective verifications of the inverse functions of binaural room impulse responses

The binaural room impulse responses (BRIRs) can be applied to 3-D sound field reconstruction, virtual reality, noise control, et al. Because the BRIRs are non-minimum phase functions, it is difficult to find the exact inverse functions of the BRIRs, especially when there are two or more sources in a reverberant space. In this work, a method was proposed to find the inverse functions of BRIRs with two sound sources in a reverberant space. The concept of time delays and the method of weighted least squares were used to find the causal, however, approximate inverse functions. The accuracy of the inverse functions was first evaluated objectively by a dummy head system. The result shows that the distortion due to crosstalk and room reverberation can be improved by 16∼18 dB. The inverse functions were also verified subjectively by 20 students. The result of subjective evaluation also shows that the inverse functions can be used successfully to reduce the crosstalk effect and the room reverberation.     

Influence of wake bubbles on optical impulse scattering echo

The model of the optical impulse scattering echo based on Mie theory and backscattering properties of bubble populations is presented. The experimental results of optical impulse scattering echo for a ship in the ocean is in agreement with the simulated experiment. The wake bubbles significantly influence the laser scattering echo process in the ocean. When the testing system is outside the wake, the distinct scattering echoes of interface between wake bubbles and water are received. When the testing system is inside the wake, the intensity of the echo attenuates with the increased scattering of wake bubbles. These phenomena of laser echoes in different periods are analyzed.     

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.     

混响场中用相关传声器识别声源直达声的近似工程法

本文研究了在混响场中用相关传声器识别主声源直达声的近似工程法，包括理论及实验技术。采用该方法对一个置于混响声污染下的恒定声源进行了直达声测量，并将测量结果与该恒定声源的消声室测量结果进行了比较，结果表明：相关传声器近似工程法在很大程度上能减少混响声对主要源声压测量的干扰，从而能在混响声场中获取一定精度的主声源声压级。     

Improving syllable identification by a preprocessing method reducing overlap-masking in reverberant environments

Overlap-masking degrades speech intelligibility in reverberation [R. H. Bolt and A. D. MacDonald, J. Acoust. Soc. Am. 21(6), 577-580 (1949)]. To reduce the effect of this degradation, steady-state suppression has been proposed as a preprocessing technique [Arai et al., Proc. Autumn Meet. Acoust. Soc. Jpn., 2001; Acoust. Sci. Tech. 23(8), 229-232 (2002)]. This technique automatically suppresses steady-state portions of speech that have more energy but are less crucial for speech perception. The present paper explores the effect of steady-state suppression on syllable identification preceded by /a/ under various reverberant conditions. In each of two perception experiments, stimuli were presented to 22 subjects with normal hearing. The stimuli consisted of mono-syllables in a carrier phrase with and without steady-state suppression and were presented under different reverberant conditions using artificial impulse responses. The results indicate that steady-state suppression statistically improves consonant identification for reverberation times of 0.7 to 1.2 s. Analysis of confusion matrices shows that identification of voiced consonants, stop and nasal consonants, and bilabial, alveolar, and velar consonants were especially improved by steady-state suppression. The steady-state suppression is demonstrated to be an effective preprocessing method for improving syllable identification by reducing the effect of overlap-masking under specific reverberant conditions.     

Localizing nearby sound sources in a classroom: binaural room impulse responses

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy.     

圆合成孔径声呐多点定位运动补偿

圆合成孔径声呐(CSAS)的成像性能受平台运动误差影响而下降,利用单侧回波可估计CSAS基阵的斜距误差,但单侧回波在小测绘带时无法估计升沉误差,针对此问题,提出了一种利用单侧回波信号的声呐平台三维运动估计和补偿方法。首先,对CSAS在不同观测角度的目标回波取极大值获得目标回波的到达时间;其次,基于多个点目标的到达时间建立CSAS三维定位模型;然后利用列文伯格-马夸尔特方法对声呐三维坐标进行估计;最后将位置估计结果与时域反投影成像方法结合实现对目标的成像.仿真结果表明:该方法能精确估计声呐平台运动误差,其空间坐标的估计误差小于仿真信号波长的1/8,从而精确补偿了CSAS在不同空间采样点上的阵元回波时间差,显著提高了目标成像质量。湖上试验结果表明,该算法能够实现对CSAS的运动误差补偿。仿真和试验结果均验证了方法的可行性和有效性。      

Pitch-based monaural segregation of reverberant speech

In everyday listening, both background noise and reverberation degrade the speech signal. Psychoacoustic evidence suggests that human speech perception under reverberant conditions relies mostly on monaural processing. While speech segregation based on periodicity has achieved considerable progress in handling additive noise, little research in monaural segregation has been devoted to reverberant scenarios. Reverberation smears the harmonic structure of speech signals, and our evaluations using a pitch-based segregation algorithm show that an increase in the room reverberation time causes degraded performance due to weakened periodicity in the target signal. We propose a two-stage monaural separation system that combines the inverse filtering of the room impulse response corresponding to target location and a pitch-based speech segregation method. As a result of the first stage, the harmonicity of a signal arriving from target direction is partially restored while signals arriving from other directions are further smeared, and this leads to improved segregation. A systematic evaluation of the system shows that the proposed system results in considerable signal-to-noise ratio gains across different conditions. Potential applications of this system include robust automatic speech recognition and hearing aid design.     

Loudness criteria in auditorium acoustics

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Blind normalization of speech from different channels

We show how to construct a channel-independent representation of speech that has propagated through a noisy reverberant channel. This is done by blindly rescaling the cepstral time series by a nonlinear function, with the form of this scale function being determined by previously encountered cepstra from that channel. The rescaled form of the time series is an invariant property of it in the following sense: It is unaffected if the time series is transformed by any time-independent invertible distortion. Because a linear channel with stationary noise and impulse response transforms cepstra in this way, the new technique can be used to remove the channel dependence of a cepstral time series. In experiments, the method achieved greater channel-independence than cepstral mean normalization, and it was comparable to the combination of cepstral mean normalization and spectral subtraction, despite the fact that no measurements of channel noise or reverberations were required (unlike spectral subtraction).     

粗晶材料晶粒散射波的实验鉴别

在利用超声波束对粗晶材料样品或工件进行检测时，接收回波中，除伤波、底波外，还存在晶粒散射波及其它干扰杂波。本文对晶粒散射波主要特征：晶粒散射波随传播时间出现的最大值位置；晶粒散射波最早到达时间；以及传播过程中晶粒散射波的频谱特性进行了实验验证。其中，用脉冲频谱合成方法，计算了双探买发．收复合声场声轴上声压分布，据此对最大值的出现位置给予了物理解释。并用短时傅立叶变换分析了声波传播过程中晶粒散射波的频谱特性。获得了一些有用结果。     

Impact localization combined with haptic feedback for touch panel applications based on the time-reversal approach

In this paper, a combined impact localization and haptic feedback system based on time-reversal signal processing is presented for touch panel applications. Theoretical impulse responses are derived from a propagation model of bending waves in a thin elastic plate. On the basis of the impulse responses, the time-reversal technique is exploited to spot the impact location as well as to generate haptic feedback. The chief advantage of the time-reversal technique lies in its robustness of tackling broadband sources in a reverberant environment. Piezoelectric ceramic plates and voice-coil motors are used as sensors for localization, whereas only voice-coil motors are used as the actuator for haptic feedback. Experimental results demonstrated that the proposed system was effective in precise impact localization for a thin panel, while haptic feedback also implemented using time-reversal principle can generate an impulse at the previously touched position. The combined impact localization and haptic feedback system is capable of enhancing the sensation of man-machine interaction in real time fashion.     

Efficient array beam forming by spatial filtering for ultrasound B-mode imaging

This paper proposes an efficient array beam-forming method using spatial matched filtering (SMF) for ultrasonic imaging. In the proposed method, ultrasonic waves are transmitted from an array subaperture with fixed transmit focus as in conventional array imaging. At receive, radio frequency echo signals from each receive channel are passed through a spatial matched filter that is constructed based on the system transmit-receive spatial impulse response. The filtered echo signals are then summed without time delays. The filter concentrates and spatially registers the echo energy from each element so that the pulse-echo impulse response of the summed output is focused with acceptably low side lobes. Analytical beam pattern analysis and simulation results using a linear array show that this spatial filtering method can improve lateral resolution and contrast-to-noise ratio as compared with conventional dynamic receive focusing (DRF) methods. Experimental results with a linear array are consistent but point out the need to address additional practical issues. Spatial filtering is equivalent to synthetic aperture methods that dynamically focus on both transmit and receive throughout the field of view. In one common example of phase aberrations, the SMF method was degraded to a degree comparable to conventional DRF methods.