Browsing by Author "Yang, Chia-Hsing"

Now showing 1 - 14 of 14

A New Design on Multi-Modal Robotic Focus Attention
(2008)
Lin, Chia-How
;
Yang, Chia-Hsing
;
Wang, Cheng-Kang
;
Song, Kai-Tai
;
Hu, Jwu-Sheng
Human detection and tracking is important for user-friendly human-robot interaction. The robot should be able to find the user autonomously and keep its attention to the user in a human-like manner. In this paper, a design and experimental study of robust human detection and tracking is presented through fusion several modalities of sensory information. The multi-modal interaction design utilizes a combination of visual, audio, and laser scanner data for reliable detection and tracking of an interested user. During tracking motion, obstacle avoidance behavior will be activated any time required to ensure safety. Furthermore, user can further assign the robot to interact with other user by speech command. Experimental results show that the robot can robustly tracks person under complex scenarios.
A robust adaptive speech enhancement system for vehicular applications
(2006-08-01)
Hu, Jwu-Sheng
;
Cheng, Chieh-Cheng
;
Liu, Wei-Han
;
Yang, Chia-Hsing
;
電控工程研究所
This work proposes and implements a novel and robust adaptive speech enhancement system, which contains both time domain and frequency domain beamformers using H-infinity filtering approach to provide a clean and undisturbed speech waveform and improve the speech recognition rate in vehicle environments. A microphone array data acquisition hardware is also designed and implemented for the proposed speech enhancement system. Mutually matched microphones are needed for traditional multidimensional noise reduction methods, but this requirement is not practical for consumer applications from the cost standpoint. To overcome this issue, the proposed system adapts the mismatch dynamics to maintain the theoretical performance allowing unmatched microphones to be used in an array. Furthermore, to achieve a high speech recognition performance, the speech recognizer is usually required to be retrained for different vehicle environments due to different noise characteristics and channel effects. The proposed system using the H-infinity filtering approach, which makes no assumptions about noise and disturbance, is robust to the modeling error in a channel recovery process. Consequently, the real vehicular experimental results show that the proposed frequency domain beamformer provides a satisfactory speech recognition performance without the need to retrain the speech recognizer.(1)
Adaptive signal blocking for generalized sidelobe canceller using matched filter array
(2008)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
電控工程研究所
;
Institute of Electrical and Control Engineering
This work proposes an adaptive beamformer based on generalized sidelobe canceller (GSC) structure with novel blocking matrix design. The classical GSC presented by Griffiths and Jim suffers from desired signal cancellation problems due to the complicated acoustic environment. This work utilizes the pseudo-inverse property of matched filters and subarray structure to design a new blocking matrix of the GSC. For practical implementation, matched filter ratios between microphone pairs are estimated, instead of estimating matched filters from the sound source to each microphone. Simulation results are presented to show the effectiveness of the proposed method.
An embedded audio-visual tracking and speech purification system on a dual-core processor platform
(2010-11-01)
Hu, Jwu-Sheng
;
Lee, Ming-Tang
;
Yang, Chia-Hsing
;
電控工程研究所
;
Institute of Electrical and Control Engineering
Design of an embedded audio-visual tracking and speech purification system is described in this paper. The system is able to perform human face tracking, voice activity detection, sound source direction estimation, and speech enhancement in real-time. Estimating the sound source directions helps to initialize the human face tracking module when the target changes the direction. The implementation architecture is based on an embedded dual-core processor, Texas Instruments DM6446 platform (Davinci), which contains an ARM core and a DSP core. For speech signal processing, an eight-channel digital microphone array is developed and the associated pre-processing and interfacing features are designed using the Altera Cyclone II FPGA. All the experiments are conducted in a real environment and the experimental results show that this system can execute all the audition and vision functions in real-time. (C) 2010 Elsevier B.V. All rights reserved.
Estimation of Sound Source Number and Directions under a Multi-source Environment
(2009)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
Wang, Cheng-Kang
;
電控工程研究所
;
Institute of Electrical and Control Engineering
Sound source localization is an important feature in robot audition. This work proposes a sound source number and directions estimation method by using the delay information of microphone array. An eigenstructure-based generalized cross correlation method is proposed to estimate time delay between microphones. Upon obtaining the time delay information, the sound source direction and velocity can be estimated by least square method. In multiple sound source case, the time delay combination among microphones is arranged such that the estimated sound speed value falls within an acceptable range. By accumulating the estimation results of sound source direction and using adaptive K-means++ algorithm, the sound source number and directions can be estimated.
Location and orientation detection of mobile robots using sound field features under complex environments
(2006)
Wu, Jwu-Sheng
;
Liu, Wei-Han
;
Cheng, Chieh-Cheng
;
Yang, Chia-Hsing
;
電控工程研究所
In this paper, the feasibility of utilizing sound for robot's pose detection is investigated, and a novel and robust robot location and orientation detection method based on sound field features for noisy environment is proposed. Unlike traditional methods, the proposed method does not explicitly consider the characteristic of direct path from sound source to microphones, nor attempt to suppress the effect of reverberations and noise signals. Instead, it utilizes the sound field features of a robot at different location and orientation in a normal environment. The sound field feature is captured by using a probability distribution estimation method called Gaussian Mixture Model (GMM). The experimental results show that this method can detect robot's location and orientation under both line-of-sight and non-line-of-sight conditions using only two microphones and is robust to environmental noise. Moreover, it can also solve the microphones' mismatch problem and can be applied to both near-field and far-field conditions. Since this method can provide global location and orientation detection, it is suitable to fuse with other localization methods to provide initial conditions for reduction of the search effort, or provide the compensation for localizing certain locations that cannot be detected using other localization methods.
Robust statistical-based speaker's location detection algorithm in a vehicular environment
(2006)
Hu, Jwu-Sheng
;
Cheng, Chieh-Cheng
;
Liu, Wei-Han
;
Yang, Chia-Hsing
;
電控工程研究所
This work proposes a threshold adaptation method to detect speaker's location in a vehicular environment. The method is robust to unmodeled sound source locations in a noisy environment and uses a single linear microphone array. The proposed approach is an improvement over previous work [12] which adopts Gaussian mixture models (GMMs) to model location-dependent but content and speaker independent acoustic characteristics of sound sources. Experimental results show that this scheme can overcome the far-filed and near-riled problem with the same architecture and perform well in both line-of-sight and non-line-of-sight cases.
Second-Order Extended H(infinity) Filter for Nonlinear Discrete-Time Systems Using Quadratic Error Matrix Approximation
(2011-07-01)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
電機工程學系
;
Department of Electrical and Computer Engineering
In this paper, the second-order extended (SOE) H(infinity) filter for nonlinear discrete-time systems is derived based on an approximation to the quadratic error matrix. The solution is obtained by the game theory approach. It is shown that the result bears a strong resemblance to the SOE Kalman filter when the performance bound goes to infinity. An example of vehicle state tracking is simulated to compare the performances of the SOE Kalman filter, the first order extended and the SOE H(infinity) filter. Noises with unknown bias are injected into both process dynamics and measurements. The results show that the SOE H(infinity) filter has the smallest state tracking error.
Second-Order Extended H-infinity Filter for Nonlinear Discrete-Time Systems Using Quadratic Error Matrix Approximation
(2011-07-01)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
電子工程學系及電子研究所
;
Department of Electronics Engineering and Institute of Electronics
In this paper, the second-order extended (SOE) H-infinity filter for nonlinear discrete-time systems is derived based on an approximation to the quadratic error matrix. The solution is obtained by the game theory approach. It is shown that the result bears a strong resemblance to the SOE Kalman filter when the performance bound goes to infinity. An example of vehicle state tracking is simulated to compare the performances of the SOE Kalman filter, the first order extended and the SOE H-infinity filter. Noises with unknown bias are injected into both process dynamics and measurements. The results show that the SOE H-infinity filter has the smallest state tracking error.
Sound Source Localization by Microphone Array on a Mobile Robot Using Eigen-Structure Based Generalized Cross Correlation
(2008)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
Wang, Cheng-Kang
;
電控工程研究所
;
Institute of Electrical and Control Engineering
Sound source localization is an important feature in robot audition. This work proposes an eigenstructure-based generalized cross correlation method for estimating time delay between microphones. From experiments, it shows that this method is more robust as compared with conventional GCC using PHAT weighting. By knowing the delays among microphones, both the sound source location and sound speed can be estimated using an array. Further, the correctness of sound speed estimation can be utilized to determine locations of multiple sources. This work derives in detail the method to estimate the directions of dual sources. A digital microphone array and associated acquisition system is developed for conducting the experiment. Experimental results prove the effectiveness of the proposed algorithms.
Sound Source Tracking and Speech Enhancement by Microphone Array on an Embedded Dual-core Processor Platform
(2008)
Hu, Jwu-Sheng
;
Huang, Chi-Yang
;
Yang, Chia-Hsing
;
Wang, Chen-Kang
;
Lee, Ming-Tang
This paper proposes and implements a robot audition module which can detect voice activity, estimate sound source direction and enhance speech signal in a noisy environment These algorithms are implemented on an embedded dual-core processor platform which includes a ARM subsystem and DSP subsystem. An eight-channel digital microphone array is also implemented for sound data acquisition. All the experiments were conducted in a real environment and the experimental results show that this system can execute all the audition functions in real-time.
Speaker attention system for mobile robots using microphone array and face tracking
(2006)
Song, Kai-Tai
;
Hu, Jwu-Sheng
;
Tsai, Chi-Yi
;
Chou, Chung-Min
;
Cheng, Chieh- Cheng
This paper presents a real-time human-robot interface system (HRIS), which processes both speech and vision information to improve the quality of communication between human and an autonomous mobile robot. The HRIS contains a real-time speech attention system and a real-time face tracking system. In the speech attention system, a microphone-array voice acquisition system has been developed to estimate the direction of speaker and purify the speaker's speech signal in a noisy environment. The developed face tracking system aims to track the speaker's face under illumination variation and react to the face motion. The proposed HRIS can provide a robot with the abilities of finding a speaker's direction, tracking the speaker's face, moving its body to the speaker, focusing its attention to the speaker who is talking to it, and purifying the speaker's speech. The experimental results show that the HRIS not only purifies speech signal with a significant performance, but also tracks a face under illumination variation in real-time.
Speech Enhancement Using Transfer Function Ratio Beamformer and Matched Filter Array
(2009)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
電控工程研究所
;
Institute of Electrical and Control Engineering
Human machine interface for speech processing often suffers from interferences which contain nonstationary signal, stationary signal and reverberation. In this paper, we consider two speech sources in the reverberant and noisy environment and we develop a speech enhancement approach to extract desired speech signal from the corrupt observations. The proposed method uses transfer function ratio beamformer, H(infinity) adaptive filter algorithm and matched filter array to perform speech enhancement as well as dereverberation at two stages. The desired speech is extracted at the first stage and the extracted speech which contains reverberation is dereverberated at the second stage.
Speech Signal Enhancement under Multiple Interferences Using Transfer Function Ratio Beamformer
(2010)
Hu, Jwu-Sheng
;
Yang, Chia-Hsing
;
電控工程研究所
;
Institute of Electrical and Control Engineering
In many practical environments, the desired speech signal is usually contaminated not only by stationary noise but also nonstationay interferences, such as competing speech. This paper proposes a speech enhancement method which can extract desired speech in a multiple interferences and reverberant environment. The proposed method uses transfer function ratio beamformer and multi-channel adaptive filter algorithm. The virtual sound source concept is proposed to simplify the theoretical treatment for multiple competing speeches. In addition, a transfer function ratio estimation method in a more practical scenario is also proposed. The experiments are performed in a real room acoustic environment.