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51st IEEE Asilomar Conference on signals Systems and Computers

作者： Xiao, Jinjun Luo, Zhi-Quan Merks, Ivo Zhang, Tao Starkey Hearing Technol Eden Prairie MN 55344 USA Chinese Univ Hong Kong Shenzhen Res Inst Big Data Shenzhen Peoples R China

ISBN: (纸本)9781538618233

In a hearing aid application, the performance of an adaptive beamformer is sensitive to errors in the acoustic transfer function or noise estimation. The estimation errors can be caused by factors including the head movement of the hearing aid users, non-stationarity of the environment, or imperfect array calibration. With such inaccurate information, the beamformer not only provides less noise reduction, but also causes undesirable speech distortions. To improve the beamformers performance in such conditions, robust beam-forming algorithms have been proposed in the literature. In particular, to minimize the speech distortions caused by imperfect look direction estimation, a robust binaural beam-forming algorithm has been proposed in [1]. This algorithm utilizes constrained optimization to improve the protection of the target speech. The beamforming adaption is based on a low-complexity iteration method called Alternating Direction Method of Multipliers (ADMM). This paper aims for evaluating the proposed binaural beamforming algorithm by comparing its performance with the Minimum Variance Distortionless Response (MVDR) beamformer in various noisy environments. The benefit of the proposed algorithm is demonstrated through both objective and subjective evaluations. In particular, multiple-microphone recordings on a pair of binaural hearing aids on a mannequin are used. Objective perception scores are calculated and compared. In addition, a subjective evaluation of speech intelligibility using normal-hearing listeners is conducted. Both the objective and subjective evaluation results show the robustness of the proposed algorithm.

关键词： microphone array signal processing acoustic beamforming binaural signal processing speech intelligibility

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A PENALIZED INEQUALITY-CONSTRAINED MINIMUM VARIANCE BEAMFORMER WITH APPLICATIONS IN HEARING AIDS

A PENALIZED INEQUALITY-CONSTRAINED MINIMUM VARIANCE BEAMFORM...

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IEEE Workshop on Applications of signal processing to Audio and Acoustics (WASPAA)

作者： Pu, Wenqiang Xiao, Jinjun Zhang, Tao Luo, Zhi-Quan Xidian Univ Natl Lab Radar Signal Proc Xian Shaanxi Peoples R China Starkey Hearing Technol Eden Prairie MN 55344 USA Chinese Univ Hong Kong Shenzhen Res Inst Big Data Shenzhen Peoples R China

ISBN: (纸本)9781538616321

A well known challenge in linearly-constrained beamforming is how to suppress multiple interferences when the degrees of freedom (DoF) provided by the array are fewer than the number of sources in the environment. In this paper, we propose a beamformer design to address this challenge. Specifically, we propose a min-max beamformer design that penalizes the spatial responses on all interferences' directions. The new design can efficiently mitigate the total interference power regardless of whether the number of interfering sources is less than the array DoF or not. We formulate this min-max beamformer design as a convex second-order cone programming (SOCP) and proposed a low complexity iterative algorithm based on alternating direction method of multipliers (ADMM) method to solve it. In the simulation, we compare the proposed beam-fomer with the linearly constrained minimum variance (LCMV) beamformer and a recently proposed inequality-constrained minimum variance (ICMV) beamformer [1]. The ability of the proposed beamformer to handle more interferences is demonstrated.

关键词： microphone array signal processing acoustic beamforming degrees of freedom binaural hearing aids

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Nonlinear speech enhancement by virtual increase of channels and maximum SNR beamformer

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EURASIP JOURNAL ON ADVANCES IN signal processing 2016年第1期2016卷 1-8页

作者： Katahira, Hiroki Ono, Nobutaka Miyabe, Shigeki Yamada, Takeshi Makino, Shoji Univ Tsukuba 1-1-1 Tennodai Tsukuba Ibaraki Japan Natl Inst Informat 2-1-2 Hitotsubashi Tokyo Japan SOKENDAI Grad Univ Adv Studies Hayama Japan

In this paper, we propose a new microphone array signal processing technique, which increases the number of microphones virtually by generating extra signal channels from real microphone signals. microphone array signal processing methods such as speech enhancement are effective for improving the quality of various speech applications such as speech recognition and voice communication systems. However, the performance of speech enhancement and other signal processing methods depends on the number of microphones. Thus, special equipment such as a multichannel A/D converter or a microphone array is needed to achieve high processing performance. Therefore, our aim was to establish a technique for improving the performance of array signal processing with a small number of microphones and, in particular, to increase the number of channels virtually by synthesizing virtual microphone signals, or extra signal channels, from two channels of microphone signals. Each virtual microphone signal is generated by interpolating a short-time Fourier transform (STFT) representation of the microphone signals. The phase and amplitude of the signal are interpolated individually. The phase is linearly interpolated on the basis of a sound propagation model, and the amplitude is nonlinearly interpolated on the basis of beta divergence. We also performed speech enhancement experiments using a maximum signal-to-noise ratio (SNR) beamformer equipped with virtual microphones and evaluated the improvement in performance upon introducing virtual microphones.

关键词： microphone array signal processing Speech enhancement Virtual microphone Maximum SNR beamformer beta divergence

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signal-Dependent Spatial Filtering Based on Weighted-Orthogonal Beamformers in the Spherical Harmonic Domain

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE processing 2016年第9期24卷 1511-1523页

作者： Delikaris-Manias, Symeon Vilkamo, Juha Pulkki, Ville Aalto Univ Sch Elect Engn Dept Signal Proc & Acoust Espoo 02150 Finland Nokia Technol Espoo 02150 Finland

Spatial filtering with microphone arrays is a technique that can be utilized to obtain the signal of a target sound source from a specific direction. Typical approaches in the field of audio underperform in practical environments with multiple sound sources and diffuse sound. In this contribution, we propose a post-filtering technique to suppress the effect of interferers and diffuse sound. The proposed technique utilizes the cross-spectral estimates of the output of two beamformers to formulate a time-frequency soft masker. The beamformers' outputs are used only for parameter estimation and not for generating an audio signal. Two sets of beamformer weights, a constant and an adaptive, are applied to the microphone array signals for the parameter estimation. The weights of the constant beamformer are designed such that they provide a spatially narrow beam pattern that is time and frequency invariant, having a unity gain toward the direction of interest. The weights of the adaptive beamformer are formulated using linearly constrained optimization with the constraint of weighted orthogonality with respect to the constant beamformer weights, as well as the unity gain toward the look direction. The orthogonality constraint provides diffuse sound suppression while the unity gain distortionless response. The cross spectrum of these two beamformers provides the target energy at a given look direction for the post filter. The study focuses on compact microphone arrays with which the typical beamforming techniques feature a tradeoff between noise amplification and spatial selectivity, especially in the low frequency region. The proposed method is evaluated with instrumental measures and listening tests under different reverberation times, in dual and multitalker scenarios. The evaluation shows that the proposed method provides a better performance when compared with a previous state-of-the-art spatial filter based on cross-pattern coherence, a linearly constrained beamformer

关键词： Beamforming cross spectrum microphone array signal processing post filtering spherical harmonics

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A ROBUST ADAPTIVE BINAURAL BEAMFORMER FOR HEARING DEVICES

A ROBUST ADAPTIVE BINAURAL BEAMFORMER FOR HEARING DEVICES

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Asilomar Conference on signals, Systems, and Computers

作者： Jinjun Xiao Zhi-Quan Luo Ivo Merks Tao Zhang Starkey Hearing Technologies Eden Prairie USA Shenzhen Research Institute of Big Data The Chinese University of Hong Kong Shenzhen China

In a hearing aid application, the performance of an adaptive beamformer is sensitive to errors in the acoustic transfer function or noise estimation. The estimation errors can be caused by factors including the head movement of the hearing aid users, non-stationarity of the environment, or imperfect array calibration. With such inaccurate information, the beamformer not only provides less noise reduction, but also causes undesirable speech distortions. To improve the beamformers performance in such conditions, robust beamforming algorithms have been proposed in the literature. In particular, to minimize the speech distortions caused by imperfect look direction estimation, a robust binaural beamforming algorithm has been proposed in [1]. This algorithm utilizes constrained optimization to improve the protection of the target speech. The beamforming adaption is based on a low-complexity iteration method called Alternating Direction Method of Multipliers (ADMM). This paper aims for evaluating the proposed binaural beamforming algorithm by comparing its performance with the Minimum Variance Distortionless Response (MVDR) beamformer in various noisy environments. The benefit of the proposed algorithm is demonstrated through both objective and subjective evaluations. In particular, multiple-microphone recordings on a pair of binaural hearing aids on a mannequin are used. Objective perception scores are calculated and compared. In addition, a subjective evaluation of speech intelligibility using normal-hearing listeners is conducted. Both the objective and subjective evaluation results show the robustness of the proposed algorithm.

关键词： microphone array signal processing Acoustic beamforming Binaural signal processing Speech intelligibility

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array microphones and signal processing within an ethernet-based AVB Network 6

Array Microphones and Signal processing within an ethernet-b...

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6th IEEE International Conference on Consumer Electronics - Berlin (ICCE-Berlin)

作者： Cohrs, Thaden Treybig, Lukas Tech Univ Ilmenau Dept Audio Visual Technol Ilmenau Germany

ISBN: (纸本)9781509020966

array microphones are frequently used in the field of speech signal applications. In this paper a use case for array microphones combined with signal processing and IEEE 802.1 AVB Ethernet-based audio signal transmission is presented. The proposed application is intended for conferencing systems. The array microphones with beam steering and sound source localization algorithm are used to span multiple speakers' recording areas. The benefit is the elimination of separate table microphones for the speakers.

关键词： microphone array signal processing audio video bridging avb audio network

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Real-time implementation of non-linear signal-dependent acoustic beamforming

Real-time implementation of non-linear signal-dependent acou...

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作者： Oliver Merilaid Aalto University

学位级别：硕士

A real-time acoustical beamforming system incorporating the cross pattern coher- ence (CroPaC) post filtering method is implemented in this thesis. The real-time implementation consists of a signal-independent beamformer that is used for spatial discrimination of a sound field. The signal of the beamformer is post filtered by modulating it with a parameter that is derived from the cross-spectrum of two directional microphone signals. The post filter is implemented to enhance perfor- mance of beamforming (increase in signal-to-noise ratio), because beamformers are not efficient in environments with high level of reverberation. The post filtering method has been previously implemented in MATLAB for non-real-time use, and this system is the first real-time implementation of an acoustical beamforming system utilizing it. The implementation is programmed in the programming language C for the graphical signal processing program Max developed by Cycling '74. It utilizes a time-frequency domain processing, and the spherical Fourier transform for a decomposition of a sound field into spheri- cal harmonic signals. The implementation can be used with microphone arrays with maximum of 32 microphone capsules, which are laid over rigid sphere with uniform or nearly-uniform arrangements. The real-time implementation can be utilized in many applications, which require algorithm to work in real-time, such as teleconferencing and acoustical cameras.

关键词： microphone array signal processing beamforming spherical harmonics post filtering

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Noise Robust Direction of Arrival Estimation for Speech Source With Weighted Bispectrum Spatial Correlation Matrix

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IEEE JOURNAL OF SELECTED TOPICS IN signal processing 2015年第5期9卷 837-851页

作者： Xue, Wei Liu, Wenju Liang, Shan Chinese Acad Sci Inst Automat Natl Lab Pattern Recognit Beijing 100190 Peoples R China

One big challenge to the robust direction of arrival (DOA) estimation for the speech source is the environmental noise. In practical conditions, the noise can be undirected or emitted from a pointed source. In order to improve the reliability of DOA estimation in various adverse noisy conditions, we propose a novel DOA estimation method in this paper, and what lies in the core in the method is the "Weighted Bispectrum Spatial Correlation Matrix (WBSCM)." The bispectrum is a kind of higher order statistics (HOS) of a signal, and the WBSCM reflects the spatial correlation of the bispectrum phase differences (BPD) between different microphones. As the HOS of the Gaussian signal is theoretically zero, by formulating in the bispectrum domain, the proposed method has an inherent advantage against the Gaussian noise. Moreover, the BPD, which is embedded in the WBSCM, contains the redundant information related to the DOA of the speech source. This redundancy helps to improve the robustness in non-Gaussian noise conditions, especially for the directional interference scenarios. In addition, the WBSCM enables bispectrum weighting to select the speech units in the bispectrum, in order to highlight the effect of these units in the DOA estimation. Similar to the signal-to-noise estimation, a decision-directed method is proposed to compute the bispectrum weights. Finally, a new DOA estimator is proposed, which is based on the eigenvalue analysis of the WBSCM. We conduct experiments under various kinds of noisy environments, and the experimental results demonstrate the effectiveness of proposed method.

关键词： Direction of arrival (DOA) estimation microphone array signal processing high order statistics bispectrum

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Perceptual Reproduction of Spatial Sound Using Loudspeaker-signal-Domain Parametrization

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE processing 2015年第10期23卷 1660-1669页

作者： Vilkamo, Juha Delikaris-Manias, Symeon Aalto Univ Sch Elect Engn Dept Signal Proc & Acoust FI-02150 Espoo Finland

Adaptive perceptual spatial sound reproduction techniques that employ a parametric model describing the properties of the sound field can reproduce spatial sound with high perceptual accuracy when compared to linear techniques. On the other hand, applying a sound-field model to control the reproduced sound may compromise the perceived quality of individual channels in cases where the model does not match the sound field. An alternative parametrization is proposed that estimates directly the perceptually relevant parameters for the target loudspeaker signals without modeling the sound field. At the synthesis stage, the loudspeaker signals with the target parametric properties are generated from the microphone signals with regularized leastsquares mixing and decorrelation. It is shown through listening experiments that the proposed method provides on average the overall perceived spatial sound reproduction quality of a state-of the-art parametric spatial sound reproduction technique, while solving the past shortcomings related to the perceived quality of the individual channels.

关键词： Least-squares mixing microphone array signal processing sound-field parametrization spatial audio reproduction

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Novel Application of Spherical microphone array Sensor with Three-Dimensional Directivity for Home and Office Environments 9

Novel Application of Spherical Microphone Array Sensor with ...

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2015 9th International Conference on Sensing Technology (ICST)

作者： Nakadai, Takahiro Yoshida, Ryuichi Taguchi, Tomoki Mizoguchi, Hiroshi Egusa, Ryohei Yamaguchi, Etsuji Inagaki, Shigenori Takeda, Yoshiaki Namatame, Miki Sugimoto, Masanori Kusunoki, Fusako Tokyo Univ Sci Dept Mech Engn Chiba Japan Kobe Univ Grad Sch Human Dev & Environm Tokyo Japan Kobe Univ Grad Sch Human Dev & Environm Kobe Hyogo Japan Tsukuba Univ Technol Fac Ind Technol Ibaraki Japan Hokkaido Univ Grad Sch Informat Sci & Technol Sapporo Hokkaido 060 Japan Tama Art Univ Dept Informat Design Tokyo Japan

ISBN: (纸本)9781479963140

Interest in sound interfaces is increasing because such interfaces do not need advanced knowledge about particular devices, nor do they require physical operation. However, such interfaces are difficult to use because the general living environment often contains background noise. In order to solve this problem, sound separation systems that use a microphone array sensor have been proposed. In previous work, this microphone array sensor was a small system that considered only two-dimensional (2-D) directivity control. In this paper, we present the design of a spherical microphone array sensor with 3D directivity. This 3-D directivity property is confirmed via an experimental evaluation.

关键词： Delay and Sum Beamforming microphone array signal processing Sound Separation System

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