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A new nonlinear model of opinion dynamics for group experts free discussion is proposed. The model takes into account the factors of similarity between the opinions, threshold of bounded confidence and trust between experts. The impacts on opinions evolution by the times of experts' speaking and rules of speech order are analyzed theoretically and compared by simulation respectively. The analytical...
Discriminative Training (DT) methods for acoustic modeling, such as MMI, MCE, and SVM, have been proved effective in speaker recognition. In this paper we propose a DT method for GMM using soft frame margin estimation. Unlike other DT methods such as MMI or MCE, the soft frame margin estimation attempts to enhance the generalization capability of GMM to unseen data in case the mismatch exists between...
Identification of non-native personnel is a critical piece of information for making crucial on-the-spot decisions for security purposes. Identification of a non-native speaker is often readily apparent in normal conversation with a native speaker through speech content and accent. Such identification which requires familiarity with language nuances may not be possible for a non-native interrogator...
Objective test methods were developed to quantify and assess the effects of personal protection equipment on auditory situational awareness. The tests utilize a custom head and torso simulator that is able to don personal protective equipment and employs acoustics, signal processing, and measurement techniques. The tests measure localization and speech intelligibility effects of personal protective...
An auditory-based feature extraction algorithm is presented. The feature is based on a recently published time-frequency transform plus a set of modules to simulate the signal processing functions in the cochlea. The feature is applied to a speaker identification task to address the acoustic mismatch problem between training and testing. Usually, the performances of acoustic models trained in clean...
An auditory-based transform is presented in this paper. Through an analysis process, the transform coverts time-domain signals into a set of filter bank output. The frequency responses and distributions of the filter bank are similar to those in the basilar membrane of the cochlea. Signal processing can be conducted in the decomposed signal domain. Through a synthesis process, the decomposed signals...
We present a USB-based, highly directional, and portable microphone array device that delivers a crisp, clear and noise-reduced speech signal. This device consists of four linearly distributed microphone sensors and a filter-and-sum beamformer designed using broadband beam-forming algorithm. The device has a narrow acoustic beam pattern and identical frequency responses for almost all speech bands...
This paper proposes a robust endpoint detection algorithm for continuous speech in noisy environment, and it can be used in automatic video caption generation systems. In the proposed algorithm, we integrate the widely used energy, zero crossing and entropy to form a new feature, EZE-feature, which possesses advantages while compensating the drawbacks of each individual. Moreover, we propose a robust...
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