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In this paper, an unsupervised adaptation algorithm for the microphone array topology of a humanoid robot is proposed, so that the spatial filtering performance is improved. In the given exemplary case, the target suppression (‘blocking’) performance of a geometrically-constrained BSS (GC-BSS) algorithm is shown to improve by the adaptation of the array topology. As a decisive feature, an online performance...
Linear microphone arrays combined with the minimum variance distortionless response (MVDR) beamformer have been widely studied in various applications to acquire desired signals and reduce the unwanted noise. Most of the existing array systems assume that the desired sources are in the broadside direction. In this paper, we study and analyze the performance of the MVDR beamformer as a function of...
In this paper, a novel speech enhancement algorithm is proposed. The algorithm controls the amount of noise reduction according to information about speech absence or presence. Compared to conventional linearly constraint minimum variance beamforming, the proposed noise reduction achieved remarkable improvement in speech recognition rates.
In this paper, we propose a new blind speech extraction microphone array combining an independent component analysis (ICA)-based noise estimator and nonlinear signal processing for achieving high-quality speech enhancement. The proposed method consists of three parts, namely, the ICA-based noise estimator for a robust target cancellation, channel-wise spectral subtraction (chSS), and post-beamforming...
In this paper, we propose a microphone array structure for a spoken-oriented robot dialog system that is designed to discriminate the direction of arrival (DOA) of the target speech and that of the robot internal noise. First, we investigate the performance of the noise estimation conducted by semi-blind source separation (SBSS) in presence of both the diffuse background noise and the robot internal...
MVDR beamformer is a robust beamforming method to enhance a desired (speech) signal in the presence of stationary noise. This paper presents a modified Subband post-filtering approach for MVDR beamformer in microphone array system. The quality of the modified Subband post-filtering is studied in simulated rooms with different noise level and is compared to wiener post-filtering proposed in the literature...
In this study, we specifically address the problem of in-vehicle voice activity detection (VAD), which has a significant importance for the speech controlled intelligent vehicle. A novel VAD system is proposed based on microphone array beam- forming and discriminative Gaussian mixture model. As a binary classification problem, the features and classifiers are explored under the in-vehicle acoustic...
Techniques of using a microphone array to determine a sound source location, the localization problem, has been studied for many years. A popular method is the so-called MUSIC (Multiple Signal Classification). There is a second type of method that tries to solve both sound separation and localization problems in one setting. The second method used for localization purpose is less known. In this study,...
In this paper, a speech enhancement algorithm based on microphone array which features the interaction between adaptive beamforming and multi-channel postfilter is proposed.A subband feedback controller based on speech presence probability is applied to Generalized Sidelobe Canceller algorithm to obtain a more robust adaptive filter in adverse environment and alleviate the problem of signal cancellation...
This paper describes a new adaptive algorithm and assesses its effectiveness within speech enhancement applications. The proposed variable step size block exact APA (VSS-BEAPA) filtering is based on the affine projection algorithm (APA) and introduces a block processing with a variable step size that allows to consider under-modeling scenarios. The algorithm shows improved convergence performance...
Speech source location estimation in a noisy, reverberant environment has attracted much attention recently. It was found that the localization method through calculating the steered response power (SRP) is more robust than time-difference-of-arrival (TDOA)-based localization method. The method places equal emphasis on each microphone pair in calculation of the SRP. In a room, each microphone is usually...
To identify the English pronunciation errors made by Chinese learners, this paper utilizes uni-directional microphones to construct a superdirective beamformer for capturing high quality input speech, and integrates the techniques of anti-model and confidence measure into the speech recognizer for accurate identification of the speaker's pronunciation errors. As to the beamformer, although designing...
This paper proposes a microphone array system separating speech signals, based on a different principle from independent component analysis (ICA). This system applies linear prediction error filters to microphone outputs, and using the prediction errors, adjusts the coefficients of adaptive filters. In this case, only the prediction errors satisfying the law of causality become available for the adjustment;...
In this paper, we propose an appropriate structure selection algorithm for less musical-noise generation in integration methods of microphone array and spectral subtraction. In our previous work, we have analyzed musical-noise reduction structure in integration methods of microphone array and spectral subtraction based on higher order statistics. However, that analysis can be applied to only Gaussian...
Compact microphone arrays allow for directional filtering with a minimum of installation space. They are therefore particularly suitable for automotive applications. Typically, compact arrays are realized as differential arrays or filter-and-sum beamformers which both show limited performance in terms of directivity. In this contribution we present a novel system for directional filtering for compact...
We present an overview of the data collection and transcription efforts for the COnversational Speech In Noisy Environments (COSINE) corpus. The corpus is a set of multi-party conversations recorded in real world environments with background noise that can be used to train noise-robust speech recognition systems. We explain the motivation for creating such a corpus and describe the resulting audio...
In this paper, we describe and review our recent development of hands-free speech dialogue system which is used for railway station guidance. In the application at the real railway station, robustness against reverberation and noise is the most essential issue for the dialogue system. To address the problem, we introduce two key techniques in our proposed hands-free system; (a) speech dialogue system...
In this paper, we conduct an analysis for reduction of musical noise in integration method of microphone array signal processing and nonlinear signal processing. In these days, for better noise reduction, integration methods of microphone array signal processing and nonlinear signal processing have been researched. However, non-linear signal processing causes musical noise. Since such musical noise...
The purpose of this study is to investigate the performance of speech presence (SP) microphone array beamforming. When the presence uncertainty of the desired speech is considered, noise reduction is greatly achieved while preserving low speech distortion level. Furthermore, we propose a novel model based speech presence probability (SPP) estimator, exploring both the sinusoid structure of speech...
This paper describes a particle filter based sound source mapping system that builds 2D sound source maps from directional sound readings taken from a mobile robot. The method uses a sound source localization model that is represented by Gaussian distribution for both direction and distance. To do this, accurate directional localization of sound sources is required, and two key components have been...
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