Vibrating intrinsic reverberation chamber is being used as an in-situ EMC test equipment for large and complex systems such as automobiles and aircrafts. In this paper, the stirring conditions, such as tightness and shaking amplitude of the walls, of a vibrating intrinsic reverberation chamber have been analyzed using the method of moments. From the viewpoint of quantitative evaluation of the flexible moving walls configuration, it was found that the random electromagnetic environment can be generated under the stirring conditions of loose configuration and a shaking amplitude more than one eighth of the wavelength at the test frequency above the lowest usable frequency.

To cope with complicated interference scenarios in realistic acoustic environment, supervised deep neural networks (DNNs) are investigated to estimate different user-defined targets. Such techniques can be broadly categorized into magnitude estimation and time-frequency mask estimation techniques. Further, the mask such as the Wiener gain can be estimated directly or derived by the estimated interference power spectral density (PSD) or the estimated signal-to-interference ratio (SIR). In this paper, we propose to incorporate the multi-task learning in DNN-based single-channel speech enhancement by using the speech presence probability (SPP) as a secondary target to assist the target estimation in the main task. The domain-specific information is shared between two tasks to learn a more generalizable representation. Since the performance of multi-task network is sensitive to the weight parameters of loss function, the homoscedastic uncertainty is introduced to adaptively learn the weights, which is proven to outperform the fixed weighting method. Simulation results show the proposed multi-task scheme improves the speech enhancement performance overall compared to the conventional single-task methods. And the joint direct mask and SPP estimation yields the best performance among all the considered techniques.

This paper presents an efficient wideband two-dimensional direction-of-arrival (DOA) estimation for an L-shaped microphone array. We propose a way to construct a wideband sample cross-correlation matrix without any process of DOA preliminary estimation, such as beamforming technique, by exploiting sample cross-correlation matrices of two different frequencies for all frequency bins. Subsequently, wideband DOAs can be estimated by using this wideband matrix along with a scheme of estimating DOA in a narrowband subspace method. Therefore, a contribution of our study is providing an alternative framework for recent narrowband subspace methods to estimating the DOA of wideband sources directly. It means that this framework enables cutting-edge techniques in the existing narrowband subspace methods to implement the wideband direction estimation for reducing the computational complexity and facilitating the estimation algorithm. Theoretical analysis and effectiveness of the proposed method are substantiated through numerical simulations and experiments, which are performed in reverberating environments. The results show that performance of the proposed method performs better than others over a range of signal-to-noise ratio with just a few microphones. All these advantages make the proposed method a powerful tool for navigation systems based on acoustic signal processing.

Aimed at long-term monitoring of daily office conversations without recording the conversational content, a system is presented for estimating acoustic nonverbal information such as utterance duration, utterance frequency, and turn-taking. The system combines a sound localization technique based on the sound energy distribution with 16 beam-forming microphone-array modules mounted in the ceiling for reducing the influence of multiple sound reflection. Furthermore, human detection using a wide field of view camera is integrated to the system for more robust speaker estimation. The system estimates the speaker for each utterance and calculates nonverbal information based on it. An evaluation analyzing data collected over ten 12-hour workdays in an office with three assigned workers showed that the system had 72% speech segmentation detection accuracy and 86% speaker identification accuracy when utterances were correctly detected. Even with false voice detection and incorrect speaker identification and even in cases where the participants frequently made noise or where seven participants had gathered together for a discussion, the order of the amount of calculated acoustic nonverbal information uttered by the participants coincided with that based on human-coded acoustic nonverbal information. Continuous analysis of communication dynamics such as dominance and conversation participation roles through nonverbal information will reveal the dynamics of a group. The main contribution of this study is to demonstrate the feasibility of unconstrained long-term monitoring of daily office activity through acoustic nonverbal information.

Reverberation chambers (RCs) are used widely in the electromagnetic measurement area. An RC is designed to have a long reverberation time, generate numerous modes, and provide good field uniformity within the chamber. The purpose of this paper is to describe the design process and measurement of the KRISS Reverberation Chamber (KRC). KRC models with 4.5m × 3.4m × 2.8m dimensions are simulated by 3D numerical simulation software. The field uniformity and correlation coefficient are then analyzed at 200MHz to obtain the optimized model. The simulation results show good performance in terms of field uniformity and are confirmed by measurement from 200MHz to 1GHz. The lowest usable frequency (LUF) of KRC was confirmed by field uniformity to be 200MHz. However, the stirrer correlation coefficient results show good performance above 300MHz.

Estimation of the time delay of arrival (TDOA) problem is important to acoustic source localization. The TDOA estimation problem is defined as finding the relative delay between several microphone signals for the direct sound. To estimate TDOA, the generalized cross-correlation (GCC) method is the most frequently used, but it has a disadvantage in terms of reverberant environments. In order to overcome this problem, the adaptive eigenvalue decomposition (AED) method has been developed, which estimates the room transfer function and finds the direct-path delay. However, the algorithm does not take into account the fact that the room transfer function is a sparse channel, and so sometimes the estimated transfer function is too dense, resulting in failure to exact direct-path and delay. In this paper, an enhanced AED algorithm that makes use of a proportionate step-size control and a direct-path constraint is proposed instead of a constant step size and the L2-norm constraint. The simulation results show that the proposed algorithm has enhanced performance as compared to both the conventional AED method and the phase-transform (PHAT) algorithm.

In our previous paper, we presented a concept of “Baseband Radio” as an ideal of future wireless communication scheme. Furthermore, for enhancing the adaptability of baseband radio, the adaptive baseband radio was discussed as the ultimate communication system; it integrates the functions of cognitive radio and software-defined radio. In this paper, two transmission schemes that take advantage of adaptive baseband radio are introduced and the results of a performance evaluation are presented. The first one is a scheme based on DSFBC for realizing higher reliability; it allows the flexible use of frequency bands over a wide range of white space. The second one is a low-power-density communication scheme with spectrum-spreading by means of frequency-domain differential coding so that the secondary system does not seriously interfere with primary-user systems that have been assigned the same frequency band.

This paper proposes a method based on modulation transfer function (MTF) to restore the power envelope of noisy reverberant speech by using a Kalman filter with linear prediction (LP). Its advantage is that it can simultaneously suppress the effects of noise and reverberation by restoring the smeared MTF without measuring room impulse responses. This scheme has two processes: power envelope subtraction and power envelope inverse filtering. In the subtraction process, the statistical properties of observation noise and driving noise for power envelope are investigated for the criteria of the Kalman filter which requires noise to be white and Gaussian. Furthermore, LP coefficients drastically affect the Kalman filter performance, and a method is developed for deriving LP coefficients from noisy reverberant speech. In the dereverberation process, an inverse filtering method is applied to remove the effects of reverberation. Objective experiments were conducted under various noisy reverberant conditions to evaluate how well the proposed Kalman filtering method based on MTF improves the signal-to-error ratio (SER) and correlation between restored power envelopes compared with conventional methods. Results showed that the proposed Kalman filtering method based on MTF can improve SER and correlation more than conventional methods.

For the realization of a high-efficiency antenna for 60GHz-band wireless personal area network, we propose placing a CMOS RF circuit and an antenna on opposing sides of a silicon chip. They are connected with low loss by a coaxial-line structure using a hole opening in the chip. Since the CMOS circuit is driven differentially, a differential-feed antenna is used. In this paper, we design and measure a differential-feed square patch antenna on a silicon chip. To enhance the radiation efficiency, it is placed on a 200µm thick resin layer. The calculated radiation efficiency of 79% includes the connection loss. A prototype antenna is measured in a reverberation chamber, and its radiation efficiency is estimated to be about 81±3%.

In a previous study, we proposed a technique to recommend candidate verbs for a method name so that developers can consistently use various verbs. In this study, we improve the rule extraction technique proposed in this previous study. Moreover, we confirm that the rank of each correct verb recommended by the new technique is higher than that by the previous technique.

This letter presents a two-stage method to extend the degenerate unmixing estimation technique (DUET) for reverberant speech separation. First, frequency-bin-wise attenuation and delay parameters are introduced and estimated by online update rules, to handle early reflections. Next, a mask reestimation algorithm based on the precedence effect is developed to detect and fix the errors on binary masks caused by late reflections. Experimental results demonstrate that the proposed method improves separation performance significantly.

Non-verbal communication incorporating visual, audio, and contextual information is important to make sense of and navigate the social world. Individuals who have trouble with social situations often have difficulty recognizing these sorts of non-verbal social signals. In this article, we propose a training tool NOCOA+ (Non-verbal COmmuniation for Autism plus) that uses utterances in visual and audio modalities in non-verbal communication training. We describe the design of NOCOA+, and further perform an experimental evaluation in which we examine its potential as a tool for computer-based training of non-verbal communication skills for people with social and communication difficulties. In a series of four experiments, we investigated 1) the effect of temporal context on the ability to recognize social signals in testing context, 2) the effect of modality of presentation of social stimulus on ability to recognize non-verbal information, 3) the correlation between autistic traits as measured by the autism spectrum quotient (AQ) and non-verbal behavior recognition skills measured by NOCOA+, 4) the effectiveness of computer-based training in improving social skills. We found that context information was helpful for recognizing non-verbal behaviors, and the effect of modality was different. The results also showed a significant relationship between the AQ communication and socialization scores and non-verbal communication skills, and that social skills were significantly improved through computer-based training.

With the rapid increase of various uses of wireless communications in modern life, the high microwave and millimeter wave frequency bands are attracting much attention. However, the existing databases on above 6GHz radio-frequency (RF) electromagnetic (EM) field exposure of biological bodies are obviously insufficient. An in-vivo research project on local and whole-body exposure of rats to RF-EM fields above 6GHz was started in Japan in 2013. This study aims to perform a dosimetric design for the whole-body-average specific absorption rates (WBA-SARs) of unconstrained rats exposed to 6GHz RF-EM fields in a reverberation chamber (RC). The required input power into the RC is clarified using a two-step evaluation method in order to achieve a target exposure level in rats. The two-step method, which incorporates the finite-difference time-domain (FDTD) numerical solutions with electric field measurements in an RC exposure system, is used as an evaluation method to determine the whole-body exposure level in the rats. In order to verify the validity of the two-step method, we use S-parameter measurements inside the RC to experimentally derive the WBA-SARs with rat-equivalent phantoms and then compare those with the FDTD-calculated ones. It was shown that the difference between the two-step method and the S-parameter measurements is within 1.63dB, which reveals the validity and usefulness of the two-step technique.

This paper proposes a method of improving the performance of blind reverberation time (RT) estimation in noisy environments. RT estimation is conducted using a maximum likelihood (ML) method based on the autocorrelation function of the linear predictive residual signal. To reduce the effect of environmental noise, a noise reduction technique is applied to the noisy speech signal. In addition, a frequency coefficient selection is performed to eliminate signal components with low signal-to-noise ratio (SNR). Experimental results confirm that the proposed method improves the accuracy of RT measures, particularly when the speech signal is corrupted by a colored noise with a narrow bandwidth.

This paper aims to achieve a high-quality exposure level quantification of whole-body average-specific absorption rates (WBA-SARs) for small animals in a medium-size reverberation chamber (RC). A two-step method, which incorporates the finite-difference time-domain (FDTD) numerical solutions with electric field measurements in an RC-type exposure system, has been used as an evaluation method to determine the whole-body exposure level in small animals. However, there is little data that quantitatively demonstrate the validity and accuracy of this method in an RC up to now. In order to clarify the validity of the two-step method, we compare the physical quantities in terms of electric field strength and WBA-SARs by using a direct numerical assessment method known as the method of moments (MoM) with ten homogenous gel phantoms placed in an RC with 2GHz exposure. The comparison results show that the relative errors between the two-step method and the MoM approach are approximately below 10%, which reveals the validity and usefulness of the two-step technique. Finally, we perform a dosimetric analysis of the WBA-SARs for anatomical mouse models with the two-step method and determine the input power related to our developed RC-exposure system to achieve a target exposure level in small animals.

Sound source localization (SSL), with a binaural input in practical environments, is a challenging task due to the effects of noise and reverberation. In psychoacoustic research field, one of the theories to explain the mechanism of human perception in such environments is the well-known equalization-cancellation (EC) model. Motivated by the EC theory, this paper investigates a binaural SSL method by integrating EC procedures into a beamforming technique. The principle idea is that the EC procedures are first utilized to eliminate the sound signal component at each candidate direction respectively; direction of sound source is then determined as the direction at which the residual energy is minimal. The EC procedures applied in the proposed method differ from those in traditional EC models, in which the interference signals in rooms are accounted in E and C operations based on limited prior known information. Experimental results demonstrate that our proposed method outperforms the traditional SSL algorithms in the presence of noise and reverberation simultaneously.

We construct two types of reverberation chambers, one is 4(m) × 2(m) × 2(m)-sized and the other is 2(m) × 2(m) × 2(m)-sized for realizing a multipath-rich environment for MIMO-OTA measurements. In this paper, we measure fundamental propagation characteristics, including amplitude statistics, multipath-delay statistics, spatial correlation characteristics, and cross polarization characteristics over a frequency range of 800MHz to 5GHz in our reverberation chamber. Also, we confirm the existence of spatially uniform area of 1m2 area, which might be sufficiently large to set a device under test (DUT) in the chamber. Theoretical considerations about the characteristics are also given to support the design of reverberation chamber.

As a first step towards the realization of high-efficiency on-chip antennas for 60GHz-band wireless personal area networks, this paper proposes the fabrication of a patch antenna placed on a 200µm thick dielectric resin and fed through a hole in a silicon chip. Despite the large tan δ of the adopted material (0.015 at 50GHz), the thick resin reduces the conductor loss at the radiating element and a radiation efficiency of 78%, which includes the connecting loss from the bottom is predicted by simulation. This calculated value is verified in the millimeter-wave band by experiments in a reverberation chamber. Six stirrers are installed, one on each wall in the chamber, to create a statistical Rayleigh environment. The manufactured prototype antenna with a test jig demonstrates the radiation efficiency of 75% in the reverberation chamber. This agrees well with the simulated value of 76%, while the statistical measurement uncertainty of our handmade reverberation chamber is calculated as ±0.14dB.

Reverberation chambers that easily create multipath-rich environments are suggested as test environments for the performance evaluation of multiple-input multiple-output (MIMO) terminals. However, the propagation environment characteristic is difficult to control in conventional reverberation chambers. In this paper, we propose an improved double-layered reverberation chamber to control the arrival wave distribution in addition to the cross-polarization power ratio (XPR). We show the design method of the double-layered reverberation chamber and the experimental results of the propagation environment control using our constructed measurement system.

This work presents a four-channel headset achieving a 5.1-channel-like hearing experience using a low-complexity head-related transfer function (HRTF) model and a simplified reverberator. The proposed down-mixing architecture enhances the sound localization capability of a headset using the HRTF and by simulating multiple sound reflections in a room using Moorer's reverberator. Since the HRTF has large memory and computation requirements, the common-acoustical-pole and zero (CAPZ) model can be used to reshape the lower-order HRTF model. From a power consumption viewpoint, the CAPZ model reduces computation complexity by approximately 40%. The subjective listening tests in this study shows that the proposed four-channel headset performs much better than stereo headphones. On the other hand, the four-channel headset that can be implemented by off-the-shelf components preserves the privacy with low cost.