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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Thi Ngoc Tho Nguyen Cagdas Tuna Shengkui Zhao Douglas L. Jones Advanced Digital Science Center (ADSC) Illinois at Singapore 138632 Singapore

ISBN: (纸本)9781509041183

When distances between microphone pairs are larger than the half-wavelength of signals, source localization methods using cross-correlation such as time-difference-of-arrival (TDOA), steered response power (SRP) are commonly used in practice. We present here a novel model that expresses microphone pairwise cross-correlations as a sum of autocorrelations of source signals shifted by the relative delays of the signals arriving at the microphone pairs, and weighted by the source power and the distances between the sources and the microphone pairs. The model is formulated as a linear inverse problem and is sparse with respect to the source power map. The source power map, which directly shows the locations of all the sound sources, can be reconstructed using l_1-norm minimization algorithms. We demonstrate the effectiveness of our model in a wildlife monitoring application, where the goal is to locate multiple frogs in a dense chorus.

关键词： cross-correlation distributed microphone array linear inverse problem multi-source source localization sparse representation

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SPATIAL-FEATURE-BASED ACOUSTIC SCENE ANALYSIS USING distributed microphone array

SPATIAL-FEATURE-BASED ACOUSTIC SCENE ANALYSIS USING DISTRIBU...

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European Signal Processing Conference

作者： Keisuke Imoto Nobutaka Ono SOKENDAI (The Graduate University for Advanced Studies) National Institute of Informatics

ISBN: (纸本)9781479988518

In this paper we propose a robust and efficient method to utilize the spatial information provided by a distributed microphone array for acoustic scene analysis. In our approach, similarly to the cepstrum, which is widely used as a spectral feature, the logarithm of the amplitude in multichannel observation is converted to a feature vector by a linear orthogonal transformation. Then, the spatial information of the acoustic scene is represented in the spatial feature space. This approach does not require the positions of the microphones and is not sensitive to the synchronization mismatch of channels, both of which make the method suitable for use with a distributed microphone array. Experimental results using real-life environmental sounds show the validity of our approach even when a smaller feature dimension than the original one is used.

关键词： Acoustic scene analysis distributed microphone array Spatial cepstrum Symmetric microphone array Isotropic sound field

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COOPERATIVE AUDIO SOURCE SEPARATION AND ENHANCEMENT USING distributed microphone arrayS AND WEARABLE DEVICES 8

COOPERATIVE AUDIO SOURCE SEPARATION AND ENHANCEMENT USING DI...

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8th IEEE International Workshop on Computational Advances in Multi-Sensor Adaptive Processing (CAMSAP)

作者： Corey, Ryan M. Skarha, Matthew D. Singer, Andrew C. Univ Illinois Urbana IL 61801 USA

ISBN: (纸本)9781728155494

Augmented listening devices such as hearing aids often perform poorly in noisy and reverberant environments with many competing sound sources. Large distributed microphone arrays can improve performance, but data from remote microphones often cannot be used for delay-constrained real-time processing. We present a cooperative audio source separation and enhancement system that leverages wearable listening devices and other microphone arrays spread around a room. The full distributed array is used to separate sound sources and estimate their statistics. Each listening device uses these statistics to design real-time binaural audio enhancement filters using its own local microphones. The system is demonstrated experimentally using 10 speech sources and 160 microphones in a large, reverberant room.

关键词： distributed microphone array audio source separation speech enhancement augmented listening hearing aids

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MULTICHANNEL FEATURE ENHANCEMENT IN distributed microphone arrayS FOR ROBUST DISTANT SPEECH RECOGNITION IN SMART ROOMS

MULTICHANNEL FEATURE ENHANCEMENT IN DISTRIBUTED MICROPHONE A...

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IEEE Workshop on Spoken Language Technology (SLT 2014)

作者： Mirsamadi, Seyedmahdad Hansen, John H. L. Univ Texas Dallas CRSS Richardson TX 75080 USA

ISBN: (纸本)9781479971299

Room reverberation and environmental noise present challenges for integration of speech recognition technology in smart room applications. We present a multichannel enhancement framework for distributed microphone arrays to mitigate the effects of both additive noise and reverberation on distant-talking microphones. The proposed approach uses techniques of nonnegative matrix and tensor factorization to achieve both noise suppression (through sparse representation of speech spectra) and dereverberation (through decomposition of magnitude spectra into convolutive components). Results of ASR experiments on the DIRHA-GRID corpus confirm that the proposed approach can achieve relative improvements of up to +20% in recognition accuracy in highly reverberant and noisy conditions using clean-trained models.

关键词： distant speech recognition distributed microphone array nonnegative matrix/tensor factorization

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AN INTEGRATION OF SOURCE LOCATION CUES FOR SPEECH CLUSTERING IN distributed microphone arrayS

AN INTEGRATION OF SOURCE LOCATION CUES FOR SPEECH CLUSTERING...

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IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Souden, Mehrez Kinoshita, Keisuke Nakatani, Tomohiro NTT Commun Sci Labs Kyoto Japan

ISBN: (纸本)9781479903566

We propose a new approach for clustering competing speech sources using distributed microphone arrays. In this approach, we first define two feature vectors where the first captures the intra-node location information while the second captures the level difference of speech energy recorded at different nodes. Then, we introduce Watson and Dirichlet mixture models to model the first and second features, respectively. We integrate both types of information in an expectation maximization algorithm to cluster the simultaneous speech sources. The performance of the proposed approach is superior to best node selection and comparable to centralized processing in terms of conventional blind source separation metrics.

关键词： distributed microphone array source clustering blind source separation expectation maximization

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distributed extended Kalman filtering for acoustic simultaneous localization and environment mapping

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APPLIED ACOUSTICS 2024年 224卷

作者： Zhao, Weiwei Chen, Zhe Yin, Fuliang Dalian Univ Technol Sch Informat & Commun Engn Dalian 116024 Peoples R China

The localization accuracy of Acoustic Simultaneous Localization and Mapping (ASLAM) often suffers from environmental noise and room reverberation. To solve this problem, an ASLAM method based on distributed extended Kalman filter is proposed. Specifically, the distance between the sound source and the robot is effectively estimated by the gating mechanism method in the case of unknown location of the sound source. Then, the Time Difference of Arrival (TDOA) between robots, the Direction of Arrival (DOA) and the distance between the sound source and the robot are presented as observation. Next, the robots' trajectories and the sound source's position are tracked by multiple extended Kalman filters. Finally, the ASLAM localization is completed by fusing the sound source and the robots' positioning information through average consensus algorithm, respectively. The proposed method can efficiently promote the ASLAM localization accuracy in complex acoustic environments. The experimental results verify the effectiveness of the proposed method.

关键词： Acoustic SLAM distributed microphone array Robot positioning Sound source localization

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An improved TF-GSC for dual-microphone interference suppression in the specific direction

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MULTIMEDIA TOOLS AND APPLICATIONS 2024年第4期83卷 11769-11783页

作者： Pang, Cong Fan, Jingjie Liang, Ruiyu Zhao, Li Cheng, Jiaming Southeast Univ Sch Informat Sci & Engn Southeast Univ Rd Nanjing 211189 Jiangsu Peoples R China Nanjing Inst Technol Sch Informat & Commun Engn Hongjing Ave Nanjing 211167 Jiangsu Peoples R China

The performance of the speech enhancement (SE) algorithm will decrease rapidly in the presence of interference, especially competing or interfering speech. In this article, an improved real-time implementation of the transfer function generalized sidelobe canceller(TF-GSC) method based on distributed dual-microphone is proposed for interference suppression in the specific direction. In our method, we first derive an improved TF-GSC method based on a primary microphone and a secondary microphone which is abbreviated as GSC-PS in the following. GSC-PS estimates the desired signal by the dual-microphone structure based on estimation of time delay of arrival and calculation of the transfer functions. After that, we propose a new adaptive interference canceller based on the multichannel speech presence probability (MC-SPP) and the output gate unit. The calculated MC-SPP is applied to the step size adjustment and cost function modification of the adaptive interference canceller, while the output gate is designed based on the normalized posterior signal-to-interference ratio difference, which is sensitive to the direction of signal sources. The simulation results show that the proposed GSC-PS algorithm outperforms the current mainstream single-channel and multi-channel SE algorithms in suppressing interference and causes less damage to the quality of the target speech. In addition, experimental results confirm the usability of the proposed algorithm in real world acoustic environment with multiple sources of noises.

关键词： Interference suppression Speech enhancement Generalized sidelobe canceller distributed microphone array

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Efficient Joint Optimization of Sampling Rate Offsets Using Entire Multichannel Signal

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2024年 32卷 1816-1828页

作者： Masuyama, Yoshiki Yamaoka, Kouei Kawamura, Takao Ono, Nobutaka Tokyo Metropolitan Univ Grad Sch Syst Design Dept Comp Sci Hino Tokyo 1910065 Japan

In this paper, we propose a joint estimation method for the sampling rate offsets (SROs) of multiple recording devices. In wireless acoustic sensor networks, distributed microphones are connected to different analog-to-digital converters, and thus SROs occur on non-reference channels, which degrades the performance of various array signal processing techniques. To address this problem, we propose to jointly estimate and compensate SROs of all the non-reference channels. Since the proposed method is formulated as a multivariate non-convex optimization problem, we derive an efficient optimization algorithm on the basis of the majorization-minimization and majorization-equalization algorithms. We further propose to update SROs with only low-frequency components in the initial iterations to avoid undesired local optima. Our experimental results validate the effectiveness of the joint estimation of all SROs and demonstrate its advantage in subsequent array signal processing.

关键词： Wireless acoustic sensor network distributed microphone array sampling rate offset auxiliary function

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Energy-Efficient Sparsity-Driven Speech Enhancement in Wireless Acoustic Sensor Networks

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IEEE-ACM TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2023年 31卷 215-228页

作者： Zhang, Jie Tao, Rui Du, Jun Dai, Li-Rong Univ Sci & Technol China Dept Elect Engn & Informat Sci Hefei 230026 Peoples R China

Wireless acoustic sensor network (WASN) has shown a superiority over conventional microphone arrays in many aspects. There exists an important tradeoff between the performance and power consumption, as usually the sensors are power driven with a limited amount of battery resource. Given a prescribed performance bound, in literature sensor selection (SS) and rate allocation (RA) methods can be leveraged to optimize the energy efficiency. In this work, we propose a joint rate allocation and sensor selection (RASS) approach to simultaneously optimize the sensor subset and rate distribution, which is formulated by minimizing the total transmission power in terms of selection and bit-rate variables and constraining the residual noise power. It can be shown that under a set of linear constraints on beamforming, the linearly-constrained minimum variance (LCMV) beamformer is the optimal noise reduction filter. Based on this, the RASS reduces to a mixed semi-definite and bilinear programming problem, which is then solved using a two-step algorithm. As the selection and bit-rate unknowns are bilinear, we first consider to optimize their product, resulting in an upper bound of RASS. Then, we use McCormick envelopes to relax the bilinear constraint, resulting in a linear program. The final selection and bit-rate solutions are obtained by posterior randomized rounding. It can be shown that SS and RA are special cases of the proposed RASS. Numerical results using simulated WASNs validate the power efficiency of the proposed method as well as the robustness against dynamic factors.

关键词： Sensors Wireless sensor networks Power demand Sensor arrays Wireless communication microphone arrays Acoustics Speech enhancement rate allocation distributed microphone array sensor selection beamformer power efficiency

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Sound Source and microphone Localization From Acoustic Impulse Responses

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IEEE SIGNAL PROCESSING LETTERS 2016年第10期23卷 1459-1463页

作者： Salvati, Daniele Drioli, Carlo Foresti, Gian Luca Univ Udine Dept Math Comp Sci & Phys I-33100 Udine Italy

This letter proposes a new method for source and microphone localization in reverberant environments using a randomly arranged sensor array, under the hypothesis that the position of one reference sensor and the geometry of the environment are known, and the other microphone positions are unknown. A minimum mean square error (MMSE) estimator that exploits early reflections is proposed. The MMSE estimator is solved by a grid search method that combines the information on early reflections estimated using a multichannel blind system identification and the time difference of arrivals between the reflections and the direct-path calculated with the image-source model. Simulations under different reverberant scenarios demonstrate the ability of the proposed approaches in localizing source and microphone.

关键词： distributed microphone array image-microphone model microphone position calibration multichannel blind system identification (BSI) sound source localization

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