記述言語：英語   掲載種別：研究論文（学術雑誌）

A complex outdoor environment includes variable illuminations, dynamic objects, and occlusion, making visual odometry in outdoor environments remains a challenging problem. This work trains a self-supervised pose estimation network based on the projection between adjacent frames. The pose between two adjacent frames is a 6D vector. It can be better solved with reliable pixels, instead of all the pixels. Thus, only reliable pixels are under consideration when training. This research proposed a boundary mask and inferior projection mask to eliminate the influence of unreliable pixels. Furthermore, we evaluated the proposed method on the KITTI datasets and compared it with other state-of-the-art deep learning-based visual odometry methods. The result shows that the proposed method can detect inferior projection and boundary pixels well. Moreover, it achieves higher accuracy in pose estimation.

DOI： 10.1007/s10015-022-00766-7

Scopus

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担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（学術雑誌）

DOI： 10.1250/ast.43.170

Scopus

CiNii Article

CiNii Research

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担当区分：最終著者   記述言語：日本語   掲載種別：記事・総説・解説・論説等（学術雑誌）

DOI： 10.20697/jasj.78.3_108

CiNii Research

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Legacy stereo music sources cause unnatural spatial impressions through the earphones and headphones reproduction due to the lack of crosstalk, which is naturally produced in loudspeakers reproduction. The amplitude-based and phase-based hybrid stereo width control method is proposed to generate the crosstalk components between channels in headphones reproduction. The control parameters for both the previously-proposed amplitude-based and hybrid methods are perceptually optimized compared with the loudspeaker reproduction. Suitable parameter values are successfully obtained for both methods with classical music sources. It is confirmed that the suitable parameter values have a different trend in those methods depending on the music sources.

Scopus

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担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Nowadays, increasing demand for teleworking needs audio capturing systems that can suppress noises and emphasize the target speech. Beamforming, which achieves spatial filtering using a microphone array, is one of the widely-used techniques for speech enhancement and noise suppression. The state-of-the-art nonlinear beamformers are designed with neural networks. In this study, the neural network with the Long Short-Term Memory as a recursive neural network is employed for dealing with acoustic signals as time series. In this study, the proposed beamformer using the neural network is optimized based on the frequency characteristics of the training data and the results of objective evaluation. The training data were prepared using an 8-ch small-scale microphone array with 10 mm microphone spacing in a soundproofed room. The experimental results show that the training data, which consist of the mixture of 1 kHz and 4 kHz sinusoidal signals, achieve suppressing noises coming from non-target directions and enhancing the target signal with low distortion.

Scopus

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記述言語：英語   掲載種別：研究論文（学術雑誌）

Numerous high-accuracy neural networks for object detection have been proposed in recent years. Creating a specific training dataset seems to be the main obstacle in putting them into practice. This paper aims to overcome this obstacle and proposes an automatic dataset creation method. This method first extracts objects from the source images and then combines them as synthetic images. These synthetic images are annotated automatically using the data flow in the process and can be used directly for training. To the best of our knowledge, this is the first automatic dataset creation method for object detection tasks. In addition, the adaptive object extraction method and created natural synthetic images make the proposed method maintain strong adaptation and generalization ability. To validate the feasibility, a dataset that includes 44 categories of objects is created for the object detection task in a vending supermarket. Under the strictest metric AP75, both the trained EfficientDet and YOLOv4 achieve higher than 95% in accuracy on the common difficulty testing set and higher than 90% in accuracy on the high difficulty testing set.

DOI： 10.1109/ACCESS.2022.3195490

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

To realize image processing as hardware module is an effective way to achieve high performance and low power processing. However, hardware development is a heavy burden for software engineer handling image processing usually. High-level synthesis, HLS, converting software program to hardware automatically is a desirable technology for software engineer with less skill of hardware design. On the other hand, the HLS has a significant problem such as it tends to generate a large and low performance hardware when the software program input has never consideration of hardware organization. This paper, focusing on color image contrast enhancement based on histogram equalization, investigates software description methods considering hardware to make HLS generate a good hardware module with high-performance and low power.

DOI： 10.1109/TENSYMP52854.2021.9550810

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（学術雑誌）

DOI： 10.12792/jiiae.9.97

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（学術雑誌）

DOI： 10.12792/jiiae.9.91

記述言語：英語   掲載種別：研究論文（学術雑誌）

Depth estimation is one of the basic and important tasks in 3D vision. Recently, many works have been done in self-supervised depth estimation based on geometric consistency between frames. However, these research works still have difficulties in ill-posed areas, such as occlusion areas and texture-less areas. This work proposed a novel self-supervised monocular depth estimation method based on occlusion mask and edge awareness to overcome these difficulties. The occlusion mask divides the image into two classes, making the training of the network more reasonable. The edge awareness loss function is designed based on the edge obtained by the traditional method, so that the method has strong robustness to various lighting conditions. Furthermore, we evaluated the proposed method on the KITTI datasets. The occlusion mask and edge awareness are both beneficial to find corresponding points in ill-posed areas.

DOI： 10.1007/s10015-021-00685-z

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Beamforming based on neural networks (NN) has been proposed as nonlinear beamforming. This method improves the performance with more microphones, but it may not be possible to easily increase the number of microphones due to hardware restrictions. Therefore, we focus on virtual observation, which is an estimation of the signal observed at a location where the microphone is not actually placed, using only actual observations. In this paper, we propose to apply virtual observation to NN-based beamforming. From the evaluation experiment, we confirmed that the proposed method can obtain sharper directivity than the case of using only actual microphones.

DOI： 10.1109/GCCE53005.2021.9621789

Scopus

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担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Beamforming has been one of the most important techniques in noise reduction under adverse acoustic environments. Traditional beamformers were designed in analytical and adaptive schemes. On the other hand, recent neural network-based beamformers achieve great success in signal enhancement. The authors have also proposed a broadband neural network-based beamformer, which can deal with wideband speech signals. The proposed beamformer employs the long short-term memory to model the temporal relationship among multichannel observations. The performance of the proposed beamformer has been confirmed with an 8-ch small-scale linear microphone array, of which the spacing between neighbouring microphones is 10 mm. It is confirmed that the proposed nonlinear beamformer could sharpen the mainlobe and decrease the sidelobes, although an analytical linear beamformer, that is, delay-and-sum beamformer, could not achieve signal enhancement with the small-scale microphone array.

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Legacy 2-ch stereo music sources cause unnatural spatial impressions through earphones and headphones reproduction due to the lack of the crosstalk produced in loudspeakers reproduction. Perceptual widths of the stereo stage in amplitude-based and phase-based stereo width control methods for headphones reproduction were matched with that of the binaural rendering which simulates loudspeakers reproduction. Paired comparison between the stereo width control methods and binaural rendering was carried out to achieve the perceptual optimization for classic, jazz, and pop music sources. Concerning the amplitude-based stereo width control method, the suitable parameter value for each sound source could be determined from the results of the listening test. It is also found that the phase-based stereo width control method should be optimized in different parameter spaces.

Scopus

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担当区分：筆頭著者,　責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Acoustic beamforming with a microphone array enables spatial filtering in a wide frequency range. It is a challenging issue to sharpen the mainlobe in the lower frequency region with a small-scale microphone array, of which the number and spacing of microphones are small. A neural network-based non-linear beamformer achieves a breakthrough in sharpening the main-lobe. The non-linear beamforming works well for the narrowband signals but is weak in wideband beamforming. The nonlinear beamforming with the long short-term memory is proposed to deal with wideband speech signals. The long short-term memory network is trained in the recurrent neural network architecture with the sequence of audio data such as speech signals. The performance of the proposed beamformer is confirmed using a small-scale 8-ch MEMS microphone array, where eight microphones are linearly arranged with the neighboring spacing of 10 mm, under a real environment. The beam-pattern of the proposed non-linear beamformer succeeds in sharpening the main-lobe although the linear delay-and-sum beamformer could not achieve frequency selectivity. The feasibility of the proposed beamformer is also confirmed in speech enhancement.

DOI： 10.3397/IN-2021-2673

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Objective measurements of speech data acquisition and presentation processes are crucial for assuring reproducibility and reusability of experimental results and acquired materials. We introduce setting and measurement examples of those conditions using an interactive and real-time acoustic measurement tool based on an extended time-stretched pulse. We also introduce supporting tools.

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

With the recent proliferation of Internet of Things (IoT) devices that can send and receive data via wireless communication, we are able to monitor and operate these devices remotely. An example of an IoT system using wireless communication is a system for anomaly detection in mechanical equipment using acoustic data. In order to detect anomalies using acoustic data, continuous recording is essential, thereby increasing the data size. Although Wi-Fi networks provide high-capacity data transfer, performance degradation cannot be avoided due to reasons such as packet losses caused by collisions with data from other devices using the same frequency and the increase in distance between two communicating devices. In the present study, we developed a wireless communication system for reliable acoustic data collection for anomaly detection in mechanical equipment. First, as preliminary experiments, we investigated the communication characteristics for the transmission of large-size data by Wi-Fi in indoor and outdoor environments. The results indicated the communication performance was insufficient for transferring all recorded data handled by this system. Therefore, we developed a simple heuristic transmission timing control method and a method that can reduce the amount of transmission data in order to realize a stable acoustic data collection system. Finally, through demonstration experiments using mechanical equipment in the field, we verified the feasibility of the acoustic data collection system.

DOI： 10.1109/CSCI54926.2021.00300

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

We introduce a new acoustic measurement method that can measure the linear time-invariant response, the nonlinear time-invariant response, and random and time-varying responses simultaneously. The method uses a set of orthogonal sequences made from a set of unit FVNs (Frequency domain variant of Velvet Noise), a new member of the TSP (Time Stretched Pulse). FVN has a unique feature that other TSP members do not. It is a high degree of design freedom that makes the proposed method possible without introducing extra equipment. We introduce two useful cases using two and four orthogonal sequences and illustrates their use using simulations and acoustic measurement examples. We developed an interactive and realtime acoustic analysis tool based on the proposed method. We made it available in an open-source repository. The proposed response analysis method is general and applies to other fields, such as auditory-feedback research and assessment of sound recording and coding.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：日本語   掲載種別：研究論文（学術雑誌）

DOI： 10.20697/jasj.76.6_343

CiNii Article

CiNii Research

その他リンク： https://ci.nii.ac.jp/naid/130007948445

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Two-channel stereophonic sound reproduction is one of the most important issues in audio engineering. Most of stereo music sources, however, cannot naturally reproduce spatial impression through earphones and headphones. In this study, stereo sources are converted by either binaural rendering or stereo width control. The former convolves head-related impulse responses into the original stereo sources, and the latter shrinks the perceptual spatial width by adding the monaural signal. Perceptual differences in between the above methods are investigated on several spatial attributes. It is confirmed that the binaural rendering is suited to reproduce the spatial impression related to the whole sound field and the stereo width control is suitable for naturally characterizing the spatial attributes for individual sound sources.

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

We propose a new family of test signals for acoustic measurements such as impulse response, nonlinearity, and the effects of background noise. The proposed family complements difficulties in existing families, the Swept-Sine (SS), pseudo-random noise such as the maximum length sequence (MLS). The proposed family uses the frequency domain variant of the Velvet noise (FVN) as its building block. An FVN is an impulse response of an all-pass filter and yields the unit impulse when convolved with the time-reversed version of itself. In this respect, FVN is a member of the time-stretched pulse (TSP) in the broadest sense. The high degree of freedom in designing an FVN opens a vast range of applications in acoustic measurement. We introduce the following applications and their specific procedures, among other possibilities. They are as follows. a) Spectrum shaping adaptive to background noise. b) Simultaneous measurement of impulse responses of multiple acoustic paths. d) Simultaneous measurement of linear and nonlinear components of an acoustic path. e) Automatic procedure for time axis alignment of the source and the receiver when they are using independent clocks in acoustic impulse response measurement. We implemented a reference measurement tool equipped with all these procedures. The MATLAB source code and related materials are open-sourced and placed in a GitHub repository.

DOI： 10.1109/APSIPAASC47483.2019.9023247

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Kyutacar

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© Proceedings of the 26th International Congress on Sound and Vibration, ICSV 2019. All rights reserved. Direction-of-arrival (DOA) estimation has been one of the important issues in acoustic signal processing. It is well known that the DOA can be estimated based on the cross correlation with stereo observations. It is, however, difficult to accurately estimate the DOA because the cross correlation-based function could not yield the sharp spatial feature. On the other hand, the multiple signal classification (MUSIC) algorithm provides the pointed spatial feature but it requires the exact number of sound sources in advance. In this paper, the cross correlation-based spatial feature is sharpened using a neural network. The neural network is trained using the cross correlation-based spatial feature calculated from the stereo observations and the binary teacher signals. It is confirmed that the non-linear cross correlation-based spatial feature has an extremely sharp peak at the direction of the target sound source without the information on the number of sound sources

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2019 The Audio Engineering Society (AES). All rights reserved. Power consumption is an important issue while listening to music using portable audio devices. The authors have previously proposed a power-saving audio playback algorithm, which has adjusted filter-bank outputs according to our auditory characteristics. It succeeds in reducing power consumption but causes perceptual distortion. In this paper, the power-saving audio playback algorithm is improved based on auditory masking, which attenuates audio components below the masking threshold. As a result of a listening test, it is confirmed that the proposed method is subjectively superior to the previous method with the same power consumption.

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Direction-of-arrival (DOA) estimation has been one of the important issues in acoustic signal processing. It is well known that the DOA can be estimated based on the cross correlation with stereo observations. It is, however, difficult to accurately estimate the DOA because the cross correlation-based function could not yield the sharp spatial feature. On the other hand, the multiple signal classification (MUSIC) algorithm provides the pointed spatial feature but it requires the exact number of sound sources in advance. In this paper, the cross correlation-based spatial feature is sharpened using a neural network. The neural network is trained using the cross correlation-based spatial feature calculated from the stereo observations and the binary teacher signals. It is confirmed that the non-linear cross correlation-based spatial feature has an extremely sharp peak at the direction of the target sound source without the information on the number of sound sources

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Beamforming has been one of the important issues in the field of multi-channel signal processing including acoustic signal processing. A wide variety of beamformers have been proposed for each application. In general, acoustic beamforming deals with broadband signals such as speech signals compared to narrowband beamforming for antenna array and radar applications. Recently, neural network-based non-linear beamformers become popular but have a problem that causes an annoying non-linear distortion on the output signal. In the case of speech enhancement, it is a serious problem because our auditory system is highly sensitive to artificial non-linear distortion on speech signals. This paper proposes to solve the problem with the relaxed dual cost functions in the neural network-based beamformer for speech enhancement. The primary cost function aims at sharpening the beam-pattern, and the second cost function is introduced to achieve decreasing speech distortion. Those cost functions are alternatively used for optimizing the beam-pattern in the frequency range of speech signals. The feasibility of the proposed method is confirmed by carrying out a listening test.

DOI： 10.18154/RWTH-CONV-239926

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Copyright © (2018) by International Institute of Acoustics & Vibration. All rights reserved. Speech intelligibility is an important factor but is partly considered for each purpose. The intelligibility of clean speech is difficult to be quantified, although the speech intelligibility indexes in noisy and reverberant environments are widely standardized. In this study, the relationship between the vowel space and subjective impressions is investigated for Japanese five vowels. The vowel space, that is the pentagon defined by the first and second formant frequencies, is considered in the Equivalent Rectangular Bandwidth scale, which is an auditory-oriented, non-linear frequency scale. Concerning the subjective evaluation, a listening test is carried out with five grade scale on subjective fluency and so on. It is found that the skewness of the vowel space is highly correlated with the subjective fluency of vowels.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2018 Audio Engineering Society. All Rights Reserved. Music appreciation can be achieved with a variety of manners such as smartphones, portable music players, car audio, and high-end audio systems. Power consumption is one of the important issues for portable electronic devices and electric vehicles. In this paper, a power saving audio playback algorithm is proposed in restraint of perceptual distortion. An original music source is passed through filterbanks and is reconstructed after increasing or decreasing the narrow-band component in each channel. The channel-dependent manipulation is carefully done not to cause perceptual distortion. The feasibility of the proposed method is evaluated by both measuring consumption current while music playback and carrying out a listening test.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© INTER-NOISE 2018 - 47th International Congress and Exposition on Noise Control Engineering: Impact of Noise Control Engineering. All rights reserved. Noise reduction has been one of the most important issues in acoustic signal processing. There are a wide variety of techniques to enhance the target signal and reduce interference. In recent days, neural network-based approaches achieve a great noise reduction performance. The authors have also proposed a neural network-based beamformer, that is, an autoencoder in the temporal domain. Channel-dependent waveforms obtained by an 8-ch linear microphone array were prepared as training data. It succeeds in forming a sharp main-lobe and suppressing side-lobes when the interference comes from the direction away from the target direction. Its performance drops when the direction of the interference is near that of the target signal. In this paper, the neural network-based beamformer is trained with amplitude spectra of the harmonic signals such as musical instruments. In other words, phase information is not used for noise reduction. The performance of the proposed method is evaluated based on signal-to-noise ratio and spectral distortion. The proposed method has an advantage in noise reduction for interferences, of which arrival directions are close to the target direction.

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：日本語   掲載種別：記事・総説・解説・論説等（その他）

担当区分：筆頭著者   記述言語：日本語   掲載種別：研究論文（学術雑誌）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2017 Institute of Noise Control Engineering. All rights reserved. Speech intelligibility under noisy environments is one of important issues in noise control engineering. It is necessary to subjectively evaluate speech intelligibility, although listening tests require enormous time and costs. Then, objective speech distortion measures have been investigated with parametric models under noisy environments. It is, however, difficult to design an all-purpose parametric distortion measure for severely distorted speech. This study aims at investigating the relationship among subjective speech intelligibility and non-parametric speech distortion measures. The non-parametric distortion measures include signal-to-noise ratio, spectral distortion, and cepstral distortion, which are calculated in each short-term frame. The spectral distortion is prepared in each octave frequency band. The short-term distortion measures yield the means and standard deviations of speech distortion. Multiple linear regression analyses are carried out, where the speech intelligibility and the objective distortion parameters are dependent and independent variables, respectively. The signal-to-noise ratio dominated speech intelligibility as expected. It is also confirmed that speech intelligibility was strongly influenced by the temporal variation of the cepstral distortion under non-stationary noise conditions.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is indispensable for human-to-machine speech interfaces, because speech signals are extremely fragile against acoustical interferences. Spatial filtering, that is, beamforming, is one of the popular solutions using multiple microphones. In the case of linear beamformers, a number of microphones are required to make the main lobe sharp at the desired direction. On the other hand, non-linear beamformers efficiently form the sharp main lobe compared with the linear beamformers. Kobatake et al. proposed a pioneering non-linear beamformer with a neural network. This method considered the narrowband optimization of the network structure, and succeeded in both achieving a sharper main lobe and decreasing grating lobes. Narrowband nonlinear beamformers have been steadily developed in the field of antenna and propagation. In this paper, a non-linear broadband beamformer with a deep neural network is proposed for speech enhancement. The proposed beamformer substitute a deep neural network for a conventional neural network, and non-equally-spaced microphone arrangement is employed for making grating lobes disappear. It is confirmed that the proposed beamformer can achieve ideal beam-pattern under a real environment.

Scopus

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担当区分：筆頭著者,　責任著者   記述言語：日本語   掲載種別：記事・総説・解説・論説等（学術雑誌）

DOI： 10.20697/jasj.73.5_297

CiNii Article

CiNii Research

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2016, German Acoustical Society (DEGA). All rights reserved. Speech intelligibility has been widely investigated for various purposes. Hands-free speech communication without any close-talking microphone becomes popular inside a vehicle for the safety reason. It is necessary to carefully install microphones so that speech signals are clearly captured under noisy in-vehicle environments. In this study, the relationship between the intelligibility of distant-talking speech and acoustical interferences is investigated inside car cabins. Target speech signals are prepared as the utterance of Japanese 4-digits random sequences. The acoustical interferences consist of exterior and interior noises. Difficulty in speech perception is defined by the speech reception threshold, which corresponds to the sound pressure level of speech at the 50 % correct score in an articulation test. It is suggested that the interior blower noise caused by an air-conditioner is the most dominant interferences in speech intelligibility. For further investigation, multiple regression analyses have been carried out using subjective indexes related to speech intelligibility. It is found that speech intelligibility can be modeled with thickness of speech, loudness of speech, and loudness of noise.

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2016 IEEE. In this paper, we propose a singing system to transmit the pseudo-vocal sound source to the vocal tract. The purpose is the quality improvement of singing voice when using an Electric-Larynx (EL), which is an alternative vocal apparatus of laryngectomee. Thus, we focus on the LPC residual wave, but it has a problem that the signal-power of the vowels /i/ and /u/ are largely reduced. Therefore we tried to improve the quality of sound by adding a signal-power compensation. We carried out listening experiments for the four sound sources «LPC residual wave», «LPC residual wave (signal-power Compensation)», «Rosenberg wave» and «EL». As a result, power-compensated LPC residual wave had the highest naturalness. However, the spectrum of the vowels /i/ differed greatly from the singing voice of the normal utterance.

DOI： 10.1109/ICCE.2016.7430662

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Recently, high resolution audio (HRA) can be played back through portable devices, and spreads across musical genres and generation. It means that most people listen to HRA through headphones and earphones. In this study, perceptual discrimination among audio formats including HRA has been investigated using headphones. 36 subjects, who have a variety of audio and musical experiences in the wide age range from twenties to seventies, participated in listening tests. Headphones presentation is superior in discriminating the details to the loudspeakers presentation. It is, however, found that the headphones presentation is weak in reproducing presence and reality. Audio enthusiasts and musicians could significantly discriminate audio formats than ordinary listeners in both headphones and loudspeakers listening conditions.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Mobile phones have caused an explosive increase in packet distribution. It causes a serious problem, that is, packet loss. Packet loss concealment is indispensable for achieving smooth speech communication. As a packet loss concealment method on a client side, a waveform substitution is popular and standardized by ITU. The ITU-T G. 711 conceals packet loss by inserting a phase-adjusted amplitude-attenuating previous packet into each break, but cannot deal with long-term breaks over 60 ms, that is, burst loss. The authors have previously proposed an alternative packet loss concealment relying on a human auditory capability. This method does not aim at restoring a waveform of the intermittent speech signal, but achieves perceptual restoration relying on phonetic restoration. When a gap of a intermittent speech signal is filled up with a loud arbitrary signal, we can smoothly listen to restored speech even if some segments of the original speech signal are completely lost. There is a trade-off between smoothness of the restored speech and noisiness of the gap-filling signal. Previously, the gap-filling signal was composed of a harmonic complex and ambient noises. In this paper, a human speech-like noise is substituted for the gap-filling signal. The speech-like noise can be prepared by repeatedly overlapping short-term human speech signals. It is confirmed that the proposed gap-filling signal succeeds in reducing its noisiness.

Scopus

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記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© Springer International Publishing Switzerland 2016. In general, aging is progressing in developed countries. Elderly people have difficulty controlling their articulation accurately due to aging. We need to improve the quality of elderly speech for smooth communication. In this paper, we analyzed the 1st formant frequency (F1) and 2nd formant frequency (F2) between the more intelligible speech and less intelligible speech of Japanese elderly people. In addition, we improved the intelligibility of less intelligible elderly speech by using the formant frequency shift method. This method is the correcting by shift value of formant frequency based on LPC. The shift value is the magnification such as expanding the F1-F2 size of less intelligible speech.

DOI： 10.1007/978-3-319-43958-7_83

Scopus

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記述言語：英語   掲載種別：研究論文（学術雑誌）

We propose a singing system to transmit the pseudo-vocal sound source into the vocal tract. In this paper, we examine the pseudo-vocal sound source that is suitable for singing system. So we carried out a comparison of LPC residual wave and Bark-LPC residual wave that the phonological component included in pseudo-vocal sound source was removed by Bark spectrum. As a result, we found that although the singing voice using Bark-LPC residual is in the low natural tendencies, the Bark-LPC residual wave can reproduce a higher speaker information contributed by the speaker characteristics.

DOI： 10.12792/jjiiae.004.01.04

CiNii Article

CiNii Research

その他リンク： https://ci.nii.ac.jp/naid/130005263509

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

© 2015 IEEE. Recent house electrical appliances generate a number of auditory signals, when a user operates an appliance or an appliance draws attention to a user. In this paper, we focus on multi-functional microwave ovens, which commonly have several auditory signals for user's operation, error occurrence, completion of cooking, and so on. A wide variety of auditory signals for current microwave ovens have two problems: it is difficult for users to image what the auditory signals inform, and the auditory signals including high frequency components are not friendly for elderly users. In this paper, popular auditory signals for microwave ovens were subjectively evaluated with/without an elderly-simulated band-pass filter, which simulated an averaged audiogram of elder people, on the five grade scales of easiness of perception, unpleasantness, and emergency. Experimental results suggest that young and elderly people might imagine different function for a single auditory signal. Finally, suitable auditory signals both for young and elderly users are proposed in the viewpoints of frequency and temporal patterns.

DOI： 10.1109/ICIEV.2015.7333977

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

High resolution audio (HRA) becomes increasingly popular both for music production and consumers. It enables to record a music performance in a wide-band and precise digital audio format. It is, however, unclear in its perceptual advantage under some listening environments. In this study, listening tests were carried out inside cars, where 34 participants listened to the same music in four different audio formats. The participants chose an audio format with better quality in paired comparison among 192 kHz/24 bits PCM, 48 kHz/16 bits PCM, and two kinds of lossy-compressed MPEG audio formats. The participants, who are familiar with HRA and live music performance, could significantly discriminate among the audio formats.

Scopus

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記述言語：日本語   掲載種別：研究論文（学術雑誌）

Utterance is very important for us to communicate with others. However, many people have speech handicaps and can't produce utterance smoothly. They mainly rely on writing and sign language for communication. But they are not easy skills for them to master. In order to help them, we developed a new utterance support system which can function through an information terminal. We made predictions which includes two additional functions of fixed phrase dictionary and the class 2-gram – one of the statistical approaches for processing natural language. Fixed phrase dictionary includes phrases often used in our daily life conversations and class 2-gram uses co-occurrence frequencies of parts of speech in Japanese. There are two purposes for us to use class 2-gram, one is to generate next candidates and the other is to change frequency. We concluded two evaluation experiments. The results show that our system needs less numbers of inputs in helping with written utterance production and using our prototype permits us to input more characters in one minute than using iPhone IME. Hereafter, we intend to create more effective dictionaries and develop new GUI to make our system more effective and faster.

DOI： 10.12792/jjiiae.003.01.02

CiNii Article

その他リンク： https://ci.nii.ac.jp/naid/130005263536

記述言語：日本語   掲載種別：研究論文（研究会，シンポジウム資料等）

日本を含む先進諸国では,高齢化が深刻な社会問題となっている.加齢は,音声コミュニケーションにも影響を及ぼす.聴覚の加齢変化については精力的に研究が行われてきたが,発話器官の加齢変化についてはまだ明らかになっていない.本稿では,発話器官の加齢変化に着目し,発話音声の静的特徴について検討する.分析に用いた音声は,64歳から87歳までの男性話者21名による日本語単語発話である.各単語の語頭母音に対して,パワースペクトルを算出し,20代話者と高齢者話者グループの平均と分散を比較した.本分析の結果,加齢により2.5kHzから4.5kHzの周波数帯域でのパワー減少と,4.5kHz以上でのパワー上昇を確認し,すべての音素において周波数特性が平坦になることが判明した.

CiNii Article

その他リンク： https://ci.nii.ac.jp/naid/110009958972

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Best Paper Award受賞

DOI： 10.1109/IS3C.2014.144

Scopus

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（学術雑誌）

DOI： 10.3390/s140609546

Scopus

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Speech quality assessment is indispensable to properly design a speech enhancement algorithm. Perceptual evaluation of speech quality (PESQ) is frequently employed as an objective speech distortion measure. The PESQ is a methodology for estimating subjective assessment of speech distortion caused by speech codecs for telephony systems. In case of noise reduction, however, a degree of speech distortion is heavier than those caused by the speech codecs. In this paper, applicability of the PESQ is investigated for noisy and noisereduced speech signals under severe noisy conditions. A relationship between PESQ scores and subjective mean opinion scores reveals that the PESQ can be applicable for heavily distorted speech only under nonstationary noisy conditions.

Scopus

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担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

A digital communication network makes our daily lives comfortable. It, however, causes a serious problem in speech communication due to a packet loss, which yields intermittent speech after decoding the packets. Packet loss concealment (PLC) is indispensable to achieve smooth speech communication over a digital network. There are various kinds of PLC methods based on wave substitution and feature prediction. Those methods aim to achieve intact restoration of a speech waveform. The ITU-T Recommendation G.711 offers a standard PLC method, but cannot conceal the packet loss, of which lost duration exceeds 60 ms. Then, G.711 PLC method is not enough to cope with the actual packet loss problem. On the other hand, the authors have proposed a perceptual restoration scheme of intermittent speech. We can listen to intermittent speech smoothly, when its gap is filled out with a loud, wide-band noise such as a white noise. This is an auditory illusion called illusory continuity, and is explained as the environmental adaptability of the human brain. In order to employ the illusory continuity for PLC, it is necessary to decrease the harshness of the loud gap-filling noise. A speech-like gap-filling signal is prepared with a mixture of less-harsh environmental noises in advance. Both dynamic and static properties of speech are incorporated into the proposed gap-filling signal. The feasibility of the proposed PLC method was examined by a listening test. Participants gave five-grade mean opinion scores on the smoothness of the restored speech, noisiness of the gap-filling signal, and the easiness of speech perception. It is confirmed that the proposed method has a significant advantage over the G.711 PLC method under long-term packet loss conditions.

Scopus

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担当区分：筆頭著者   記述言語：日本語   掲載種別：研究論文（研究会，シンポジウム資料等）

高齢化は大きな社会問題であり、加齢による音声コミュニケーションに生じる障害も切実な問題である。加齢による聴力低下は、60歳代以前にも見受けられるが、70歳を超えると急激に高周波数成分の聴力閾値が上昇する。70歳代では、聴覚障害の自覚がなくても、加齢による高音障害型感音性難聴と診断されることが多い。これにより、子音が聴き取りづらくなったり、雑音のなかでの音声の聞き取り能力が低下する。一方、音声知覚において、我々は、声からある程度の年齢を推定できる。つまり、我々は、聴覚的印象として、高齢者音声らしさを表す複数の特徴を知っている。その一つが嗄声であり、これは高齢者音声を特徴付ける聴感印象として数多く報告されている。著者は、高周波数帯域における発話エネルギー上昇が、嗄声の音響的特徴であることを確認した。従来の研究では、高齢者の聴力特性と音声の音響的特徴は、それぞれ独立して調査されてきた。本稿では、「ことばの鎖」を拠り所として、加齢による聴力低下と発話音声変形の相互作用について総合的に検討する。

CiNii Article

その他リンク： https://ci.nii.ac.jp/naid/110009783062

記述言語：英語   掲載種別：研究論文（学術雑誌）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Demand for evaluating speech quality is on the increase. It is advisable for evaluating the speech quality to employ the common objective measure for the wide variety of adverse speech signals. Unfortunately, current speech quality measures do not suit for heavily distorted speech signals. In this paper, both the applicability and the limit of the perceptual evaluation of speech quality (PESQ) are investigated compared with the subjective mean opinion score (MOS) for noise-added and noise-reduced speech signals. It is found that the PEAQs are compatible with the MOSs for the noise-reduced speech signals in the non-stationary noise conditions.

Scopus

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担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

DOI： 10.1121/1.4799431

Scopus

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

It is important for speech applications to quantify the intelligibility and the quality of speech. There are some means of evaluating the intelligibility and the quality objectively and subjectively. The objective evaluation is economical, but the general-use distortion measure is not available. On the other hand, the subjective evaluation requires a cost-consuming formal listening test. The right evaluation should be carried out in the right purpose. No almighty methodology has been established at the moment. In this paper, brain activities involved in evaluating the noisy and noise-reduced speech are measured using the functional magnetic resonance imaging (fMRI) technique. It is found that the brain activity in evaluating the noise-reduced speech varies depending on the noise reduction approach. The noise-reduced signals, which are obtained by the linear processing such as the delay- And-sum beamforming, calm down the brain activity compared with the noisy speech signals. On the other hands, non-linear signal processing such as the spectral subtraction activates the auditory cortex intensely. It is necessary for designing the advanced speech distortion measure to independently consider how the noise is removed and how the speech is distorted by noise reduction.

Scopus

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記述言語：日本語   掲載種別：研究論文（学術雑誌）

In this paper, sound quality of high resolution audio (HRA) is investigated in the view point of auditory perception. Perceptual characteristics of HRA have been examined by listening tests compared with the standard audio CD and the compressed MPEG audio layer-3 (MP3) qualities. The listening tests were carried out by the paired comparison, and the participants were asked to discriminate among HRA, CD, and MP3. Jazz music was selected out as the original HRA. Both CD and MP3 quality audio files were prepared from the original HRA file by the authors. It is found that the participants prefer the audio format with richer information, that is, HRA is most desirable and CD is better than MP3. The questionnaires filled in by participants indicate that subjective characteristics of HRA include much presence, high spatial clarity, and wide spaciousness. Additionally, perceptual discrimination in quantization distortion is also focused on in between 16 bits and 24 bits resolution. As the result of the listening test with 28 participants, the difference of the quantization resolution was discriminated with the accuracy of 60.3 %. Concerning the perceptual discrimination of HRA, quantization resolution is an important factor as well as its wide frequency range.

DOI： 10.12792/jjiiae.001.02.003

Kyutacar

CiNii Article

その他リンク： https://ci.nii.ac.jp/naid/130005263583

担当区分：筆頭著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Recently, a new obstruction to speech communication, that is, the short-term intermittence of the speech signal, is occurred due to the congestion in the digital communication network. Mobile phones and smart phones have spread rapidly, and then the communication lines are occupied by the heavy traffic of the received/transmitted packets. Therefore, the packet associated with speech can also be lost in the congested communication line. It is not advisable to require the re-transmission of the lost packet in speech communication, because the re-transmission of the lost packet disturbs smooth speech communication and causes the further traffic in the communication line. The restoration of the intermittent speech should be completed on the user's client equipment. The copy of the previous packet should be the easiest approach in restoring the intermittent speech, although the restored speech must be perceived slowly and lazily. The authors have proposed a novel approach to the restoration of the intermittent speech relying on the human auditory capability [Proc. ICSV19, 2012]. This is not an exact physical restoration of the intermittent speech signal, but a perceptual restoration. When the gap of the intermittent speech signal is filled up with an extremely noisy noise signal, we can listen to the intermittent speech smoothly even if some segments of the original speech signal are lost completely. The authors have proposed a practical scheme of the restoration based on our phonemic restoration capabilities. The previously proposed method has employed the less harsh noise and the speech-like harmonic component designed in the frequency domain as the gap-filling signal. Furthermore, in this paper, the temporal characteristics of speech are introduced into the gap-filling signal. It is confirmed that the amplitude modulation and the temporal decay in the gap-filling signal improve the auditory impression of the restored speech signals.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84897035542&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

It is indispensable for designing a noise reduction algorithm to evaluate the quality of the noise-reduced speech signals accurately and efficiently. Subjective evaluation is an ideal approach, but requires a time-consuming listening test with a number of subjects. Then, objective distortion measures are often employed for efficient evaluation. However, almost all the distortion measures do not consider the temporal variation of the speech distortion. In this paper, the temporal aspect of the short-term speech distortion is investigated based on higher-order statistics such as variance, skewness, and kurtosis for short-term segmental SNR distribution. It is interesting that the skewness of the segmental SNR gives a good explanation for the discrepancy between subjective and objective evaluation.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84883590418&origin=inward

担当区分：責任著者   記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is indispensable for designing high-quality speech interfaces. Various kinds of beamformers have been proposed for speech enhancement by spatial filtering. Delay-and-sum beamforming is usually expensive for speech enhancement with numbers of microphones, but provides less-distorted speech signals due to linear signal processing. Null-steering subtractive beamforming achieves efficient speech enhancement, but resultant speech signals include artificial distortion by non-linear signal processing. Each beamformer has both an advantage and a drawback. Directionof- arrival (DOA) is the most important spatial feature for beamforming. The proposed versatile beamforming aims at integrating delay-and-sum beamforming with null-steering beamforming adaptively under time-varying acoustic environments. Confidence for each DOA estimate can be estimated based on a spatial feature, e.g., a cross-correlation function for stereo observations, and is the key for achieving the adaptive integration. Null-steering beamforming can be employed with a reliable DOA estimate, and delay-and-sum beamforming should be used with an unreliable DOA estimate. The DOA confidence varies time by time under non-stationary acoustic environments. The propose versatile beamformer chooses between delay-and-sum and null-steering subtractive beamformers relying on the confidence of each DOA confidence in non-stationary acoustic environments.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84866005187&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is indispensable for robust speech processing in noisy environments. It is, however, difficult to achieve effective speech enhancement under practical, non-stationary, and unknown noisy environments. Challenge to design an almighty algorithm is not realistic for speech enhancement in real environments. In other words, the right scheme should be prepared for speech enhancement in the right noisy environment. In this paper, a novel approach is introduced to achieve robust speech enhancement. Provided that two or more microphones are available, we can estimate a direction of arrival (DOA) from a spatial feature, which can be also used for spatial filtering as a means of speech enhancement. The author has already proposed to give reliability, which is also estimated from the spatial feature, to each estimate of the DOA. The proposed method achieves speech enhancement by adaptively selecting the suitable approach relying on the reliability of each DOA estimate. Beamforming and spectral subtraction are prepared as candidates for spatial and spectral filtering, respectively. Each output is merged with the time-varying mixing ratio depending on the reliability of each DOA estimate. Then, the proposed method can deal with non-stationary noisy environments by introducing the adaptive integration between spatial and spectral filtering. Feasibility of the proposed idea is confirmed under a less-reverberant, non-stationary noisy condition. Copyright © (2011) by the Institute of Noise Control Engineering.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84867960160&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

It is important for designing a noise reduction algorithm to evaluate the quality of the noise-reduced speech signals accurately and efficiently. Subjective evaluation is ideal, but requires a time-consuming listening test with a lot of subjects. Then, objective distortion measures are often used for efficient evaluation. However, almost all the distortion measures do not consider the temporal variation of the speech distortion. In this paper, the temporal aspect of the short-term speech distortion is investigated based on higher-order statistics, that is, variance, skewness, and kurtosis. It is interesting that the skewness of the short-term SNR gives a good explanation for the discrepancy between subjective and objective evaluation.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84866309724&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

This paper proposes to estimate the reliability of each direction-of-arrival (DOA) estimate explicitly. Observed spatial features, which give DOA estimates, are usually distorted by acoustical noise and reverberation. Then, distorted spatial features are filtered out by particle filters with a system model, which describes sound source dynamics on spatial state space. After particle filtering is carried out, set of the weighted particles can be considered as the discrete approximation of the true spatial feature. It is reasonable that particles with large weights concentrate around the true DOA on spatial state space. In this paper, crest factor, skewness, and kurtosis are calculated based on the second, third, and forth order moments from the particle set for representing how particles concentrate simply. It is confirmed that the skewness confidence measure is the most suitable for estimating DOA reliability under both directional and distributed noise conditions. © EURASIP, 2010.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84863793327&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Speech enhancement is essential for many audio applications with speech input. Those devices are supposed to be used under a wide variety of acoustic conditions. Then, speech enhancement should be effective even in unknown, time-varying, and non-stationary noisy environments. The author has proposed an efficient noise reduction method, which aims at making the most use of stochastic spatial information [Mizumachi, Proc. ICSV16, 2009]. The spatial and acoustical features have been prepared based on directions of sound sources and spectral characteristics of a target and acoustical interferences, respectively. In this paper, those stochastic features are extended including much richer spatial information. New spatial features are derived from higher order statistics of the spatial information extracted from the 2-ch observations. The proposed method can deal with non-stationary noisy environments, because it takes the suitable scheme in noise reduction relying on the spatial and acoustical features flexibly. In this paper, the feasibility of the proposed method is investigated under non-stationary noisy environments.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84871439200&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Noise reduction has been one of important issues in acoustic signal processing. Until now, various kinds of algorithms have been proposed to achieve noise reduction, but we have not established an almighty algorithm, which always works well in the real world. Each algorithm has both advantage and disadvantage in noise reduction. In this paper, a practical approach to noise reduction is proposed by introducing an idea of self-organization for determining the suitable algorithm in time-varying acoustic scenes based on acoustic and spatial features. Time evolution of acoustic and spatial features and reliability of direction-of-arrival estimates are used for selecting the suitable approach out of spectral subtraction with acoustic features and beamforming with spatial features. Feasibility of the self-organized approach has been confirmed under non-stationary noise environments.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84868667154&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

This paper proposes to estimate the reliability of each direction-of-arrival (DOA) estimate explicitly. Observed spatial features, which give DOA estimates, are usually distorted by acoustical noise and reverberation. Then, distorted spatial features are filtered out by particle filters with a system model, which describes sound source dynamics on spatial state space. After particle filtering is carried out, set of the weighted particles can be considered as the discrete approximation of the true spatial feature. It is reasonable that particles with large weights concentrate around the true DOA on spatial state space. In this paper, crest factor, skewness, and kurtosis are calculated based on the second, third, and forth order moments from the particle set for representing how particles concentrate simply. It is confirmed that the skewness confidence measure is the most suitable for estimating DOA reliability under both directional and distributed noise conditions. © 2010 TSI Press.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=78651417406&origin=inward

記述言語：英語   掲載種別：研究論文（国際会議プロシーディングス）

Separation of sound source can be achieved by spatial filtering with multi-channel acoustic observations. However, the right algorithm should be prepared in each condition of acoustic scene. It is difficult to provide the suitable algorithm under real acoustic environments. In this paper, an adaptive source separation scheme is proposed based on the reliability of a spatial feature, which gives an estimate of direction of arrival (DOA). As confidence measures for DOA estimates, the third and fourth moments for spatial features are employed to measure how sharp the main-lobes of spatial features are. This paper proposes to selectively use either spatial filters or frequency-selective filters without spatial filtering depending on the reliability of each DOA estimate.

Scopus

その他リンク： https://www.scopus.com/inward/record.uri?partnerID=HzOxMe3b&scp=84866026425&origin=inward

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開催期間： 2009年11月   記述言語：日本語  

開催期間： 2009年10月   記述言語：英語  

開催期間： 2009年09月   記述言語：日本語  

開催期間： 2009年09月   記述言語：日本語  

開催期間： 2009年09月   記述言語：日本語  

開催期間： 2009年09月   記述言語：日本語  

開催期間： 2009年03月   記述言語：日本語  

開催期間： 2009年03月   記述言語：日本語  

開催期間： 2009年03月   記述言語：日本語  

開催期間： 2008年11月   記述言語：日本語  

開催期間： 2008年10月   記述言語：日本語  

開催期間： 2008年10月   記述言語：英語  

開催期間： 2008年09月   記述言語：日本語  

開催期間： 2008年09月   記述言語：日本語  

開催期間： 2008年09月   記述言語：日本語  

開催期間： 2008年09月   記述言語：日本語  

開催期間： 2008年03月   記述言語：日本語  

開催期間： 2007年10月   記述言語：日本語  

開催期間： 2007年10月   記述言語：日本語  

開催期間： 2007年09月   記述言語：日本語  

開催期間： 2007年09月   記述言語：日本語  

開催期間： 2007年09月   記述言語：日本語  

開催期間： 2007年03月   記述言語：日本語  

開催期間： 2007年03月   記述言語：日本語  

開催期間： 2007年01月   記述言語：日本語  

開催期間： 2006年11月28日 - 2006年12月02日   記述言語：英語  

開催期間： 2006年09月   記述言語：日本語  

開催期間： 2006年09月   記述言語：日本語  

開催期間： 2006年09月   記述言語：日本語  

開催期間： 2006年09月   記述言語：日本語  

開催期間： 2006年03月   記述言語：日本語  

開催期間： 2005年12月   記述言語：日本語  

開催期間： 2005年09月   記述言語：日本語  

開催期間： 2005年09月   記述言語：日本語  

開催期間： 2005年09月   記述言語：日本語  

開催期間： 2005年09月   記述言語：日本語  

開催期間： 2005年03月   記述言語：日本語  

開催期間： 2005年03月   記述言語：日本語  

開催期間： 2005年02月   記述言語：日本語  

開催期間： 2004年12月   記述言語：日本語  

開催期間： 2004年12月   記述言語：日本語  

開催期間： 2004年10月   記述言語：日本語  

開催期間： 2004年07月   記述言語：日本語  

公開番号：'特許公開２００５－２６０７４４  公開日：2005年09月22日