Browsing by Author "Bai, Mingsian R."

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A study of fault diagnosis in a scooter using adaptive order tracking technique and neural network
(2009-01-01)
Wu, Jian-Da
;
Wang, Yu-Hsuan
;
Chiang, Peng-Hsin
;
Bai, Mingsian R.
;
機械工程學系
An expert system for scooter fault diagnosis using sound emission signals based on adaptive order tracking and neural networks is presented in this paper. The order tracking technique is one of the important approaches for fault diagnosis in rotating machinery. The different faults present different order. figures and they can be used to determine the fault in mechanical systems. However, many breakdowns are hard to classify correctly by human experience in fault diagnosis. In the present study, the order tracking problem is treated as a parametric identification and the artificial neural network technique for classifying faults. First, the adaptive order tracking extract the order features as input for neural network in the proposed system. The neural networks are used to develop the training module and testing module. The artificial neural network techniques using a back-propagation network and a radial basis function network are proposed to develop the artificial neural network for fault diagnosis system. The performance of two techniques are evaluated and compared through experimental investigation. The experimental results indicated that the proposed system is effective for fault diagnosis under various engine conditions. (C) 2007 Elsevier Ltd. All rights reserved.
Acoustic camera
(2010-10-28)
Bai, Mingsian R.
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Lin, Jia-Hong
;
Liu, Kuan-Liang
An acoustic camera comprises a first sound pick-up device, a second sound pick-up device, and a switch. The switch is respectively connected to the first sound pick-up device and the second pick-up device and used to select the first sound pick-up device or the second sound pick-up device to reconstruct the sound field of the sound source of a detected object. The first sound pick-up device has a first microphone array, and the first microphone array is a near-field uniform microphone array. The second sound pick-up device has a second microphone array, and the second microphone array is a far-field non-uniform microphone array.
Acoustical Design of a Bluetooth Earphone Using Simulated Annealing Optimization
(2010-07-01)
Bai, Mingsian R.
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Kuo, Yu-Chih
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機械工程學系
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Department of Mechanical Engineering
Faced with various conflicting issues arising from sensitivity, bandwidth, and miniaturization requirements for Bluetooth earphones, it is desirable to develop a systematic way to attain a design that would meet these requirements. Based on a lumped-parameter model, the design of the earphone enclosure is optimized using the simulated annealing (SA) algorithm. Experimental results obtained using a type 3.3 ear simulator, the IEC 60318-4 (which supersedes IEC 60711 in 2010), reveal that the optimized design has resulted in a significant performance enhancement, complying with the frequency response mask dictated in the 3GPP2 standard.
An expert system for the diagnosis of faults in rotating machinery using adaptive order-tracking algorithm
(2009-04-01)
Wu, Jian-Da
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Bai, Mingsian R.
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Su, Ju-Cheng
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Huang, Chin-Wei
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機械工程學系
This paper describes an application of an adaptive order-tracking technique for the diagnosis of faults in rotating machinery. Conventional methods of order-tracking are primarily based on Fourier analysis with reference to shaft speed. Unfortunately, in some applications of order-tracking performance is limited, such as when a smearing problem arises and also in a multiple independent shaft system. In this study, the proposed fault diagnostic system is based on a recursive least-square (RLS) filtering algorithm. The problem is treated as the tracking of various frequency bandpass signals. Order amplitudes can be calculated with high-resolution in real-time implementation. The algorithm is implemented on a digital signal processor (DSP) platform for diagnosis and evaluated by experimental investigation. An experimental investigation is implemented to evaluate the proposed system in two applications of gear-set defect diagnosis and in the diagnosis of damaged engine turbocharger blades. The results of the experiments indicate that the proposed algorithm is effective in fault diagnosis for both experimental cases. Furthermore, a characteristic analysis and experimental comparison of a vibration signal and a sound emission signal for the present algorithm are also presented in this report. (C) 2008 Elsevier Ltd. All rights reserved.
An optimized energy harvester for moving mass induced vibration
(2011-05-01)
Bai, Mingsian R.
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Tsai, Yao Kun
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機械工程學系
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Department of Mechanical Engineering
This paper presents a piezoelectric energy harvester by which the vibration energy induced by a moving mass is converted to electrical energy through the piezoelectric effect. An electromechanically coupled finite element model (FEM) based on the Euler-Bernoulli beam theory is employed to estimate the electrical energy that can be generated by the energy harvester. The effects of mass ratio, beam length, travel time and load resistance on the energy output are examined. Experiments are conducted to verify the numerical model. The experimental results are in good agreement with the numerical prediction. In the design stage, the nonlinear conjugate gradient (CG) algorithm is applied for the calculation to maximize the energy throughput from the energy harvester. Results have shown that the harvested energy depends heavily upon the optimal choice of load resistance and travel time of the moving mass. In addition, the longer the beam or the higher the mass ratio, the higher the energy throughput that can be achieved.
Comparative study of audio spatializers for dual-loudspeaker mobile phones
(2007-01-01)
Bai, Mingsian R.
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Shih, Geng-Yu
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Lee, Chih-Chung
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機械工程學系
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Department of Mechanical Engineering
MPEG-1, layer 3 handsets equipped with dual loudspeakers and three-dimensional audio modules have received much attention in the market of consumer electronics. To create spatial impression during audio reproduction, the head-related transfer function (HRTF) and the crosstalk cancellation system (CCS) are key elements in many audio spatializers. However, there are many factors that one should take into account during the design and implementation stages of an audio spatializer in the handset application. In the paper, a comprehensive study was undertaken to compare various audio spatializers for use with dual- loudspeaker handsets, in the context of inverse filtering strategies. Two deconvolution approaches, the frequency-domain method and the time-domain method, are employed to design the required inverse filters. Different approaches to design audio spatializers with the HRTF, CCS, and their combination are compared. In particular, two modified CCS approaches are suggested. Issues in the implementation phase such as regularization, complex smoothing, and structures of inverse filters are also addressed in the paper. Comprehensive objective and subjective tests were conducted to investigate the aforementioned aspects of audio spatializers. The data obtained from the subjective tests are processed by using the multianalysis of variance to justify statistical significance of the results. (c) 2007 Acoustical Society of America.
Comparative Study of Design and Implementation Strategies of Automotive Virtual Surround Audio Systems
(2010-03-01)
Bai, Mingsian R.
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Lee, Chih-Chung
;
機械工程學系
;
Department of Mechanical Engineering
A comprehensive study was conducted to compare various design and implementation strategies of automotive virtual surround audio. Nowadays there is increased proliferation of multichannel audiovisual systems used in cars. However, the interior of a car is known as a notorious listening environment due to reflections in a confined space. nonideal user/loudspeaker positions, and ambient noise. It is then desirable to develop audio systems that are capable of rendering quality spatial sound fields in harsh car environments. Four design approaches are proposed for two-channel and 5.1-channel input. For two-channel input a method of reverberation-based upmixing with inverse filtering and another method of up/down mixing with weighting and delay are presented. The upmixing algorithm is used to convert two-channel signals to four-channel signals, while the downmixing algorithm does just the opposite. Inverse filters are employed to position the virtual sound images according to the 5.1 configuration. For 5.1-channel input a method of downmixing with inverse filtering and another method of downmixing with weighting and delay are presented. These processing algorithms have been practically implemented on a car. Simulations and experiments were conducted for validating the proposed spatial audio systems. Subjective listening tests were also conducted to compare these methods with the data processed by multivariate analysis of variance (MANOVA). The experimental results reveal that inverse filtering is most effective for single-passenger mode. especially at rear seats. However, if there is more than one passenger, the alternative methods may be more appropriate in terms of rendering performance and computational complexity.
Design and implementation of a hybrid sub-band acoustic echo canceller (AEC)
(2009-04-10)
Bai, Mingsian R.
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Yang, Cheng-Ken
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Hur, Ker-Nan
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機械工程學系
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Department of Mechanical Engineering
An efficient method is presented for implementing an acoustic echo canceller (AEC) that makes use of hybrid sub-band approach. The hybrid system is comprised of a fixed processor and an adaptive filter in each sub-band. The AEC aims at reducing the echo resulting from the acoustic feedback in loudspeaker-endosure-microphone (LEM) systems such as teleconferencing and hands-free systems. In order to cancel the acoustical echo efficiently, various processing architectures including fixed filters, hybrid processors, and sub-band structure are investigated. A double-talk detector is incorporated into the proposed AEC to prevent the adaptive filter from diverging in double-talk situations. A de-correlation filter is also used alongside sub-band processing in order to enhance the performance and efficiency of AEC. All algorithms are implemented and verified on the platform of a fixed-point digital signal processor (DSP). The AECs are evaluated in terms of cancellation performance and computation complexity. In addition, listening tests are conducted to assess the subjective performance of the AECs. From the results, the proposed hybrid sub-band AEC was found to be the most effective among all methods in terms of echo reduction and timbral quality. (C) 2008 Elsevier Ltd. All rights reserved.
Development of an expert system for fault diagnosis in scooter engine platform using fuzzy-logic inference
(2007-11-01)
Wu, Jian-Da
;
Wang, Yu-Hsuan
;
Bai, Mingsian R.
;
機械工程學系
;
Department of Mechanical Engineering
In the present study, a fault diagnosis system using acoustic emission with an adaptive order tracking technique and fuzzy-logic interference for a scooter platform is described. Order tracking of acoustic or vibration signal is a well-known technique that can be used for fault diagnosis of rotating machinery. Unfortunately, most of the conventional order-tracking methods are primarily based on Fourier analysis with the revolution of the machinery. Thus, the frequency smearing effect often arises in some critical conditions. In the present study, the order tracking problem is treated as the tracking of frequency-varying bandpass signals and the order amplitudes can be calculated with high resolution. The order amplitude figures are then used for creating the data bank in the proposed intelligent fault diagnosis system. A fuzzy-logic inference is proposed to develop the diagnostic rules of the data base in the present fault diagnosis system. The experimental works are carried to evaluate the effect of the proposed system for fault diagnosis in a scooter platform under various operation conditions. The experimental results indicated that the proposed expert system is effective for increasing accuracy in fault diagnosis of scooters. (c) 2006 Elsevier Ltd. All rights reserved.
Device and method for integrating sound effect processing and active noise control
(2007-05-31)
Bai, Mingsian R.
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Ou, Kuen-Ying
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Lin, Jain-Liang
A device and a method for integrating 3D sound effect processing and active noise control are proposed. A digital signal processor incorporates an artificial reverberator and a 3D spatial audio processor into an audio module. The audio signal is presented via an earphone. Next, a microphone embedded in the vicinity of the loudspeaker inside the headset is used to sense an external noise while playing, and feed it back to an active noise controller, which generates an anti-noise to eliminate the external noise. Therefore, the signal to noise ratio can be increased and the 3D sound field effect can be significantly enhanced. In addition, a head-related transfer function is more efficiently implemented on the basis of an interaural transfer function in the spatial audio processing to reduce the filter order lower and hence the computation loading.
Electroacoustic analysis of an electret loudspeaker using combined finite-element and lumped-parameter models
(2009-06-01)
Bai, Mingsian R.
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Chen, Rong-Liang
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Wang, Chun-Jen
;
機械工程學系
;
Department of Mechanical Engineering
An unconventional type of electrostatic loudspeaker is presented in this paper. The loudspeaker made of thin, light, and flexible electret material lends itself well to the space-concerned applications. Electrical impedance measurement reveals that the coupling between the electrical system and the mechanical system is weak, which renders conventional parameter identification based on electrical impedance measurement impractical. A different approach is thus employed to model the electret loudspeaker. To predict the loudspeaker's dynamic response, finite-element analysis (FEA) is conducted on the basis of a simple model and a full model. In the simple model, FEA is applied to model the electret membrane, leaving the rest of system as rigid parts. In the full model, FEA is applied to model the entire membrane-spacer-back plate assembly. Velocity response of the membrane subject to a uniformly distributed force is calculated using FEA harmonic analysis. Mechanical impedance is then calculated with the velocity response. The acoustical impedance due to the back cavity, pores, and the radiation loading at the front side is calculated by theoretical formulas. The volume velocity of the membrane and the resulting on-axis sound pressure level are predicted with electrical-mechanical-acoustical analogous circuits. The response data predicted by the simulation compare very well with experimental measurements. (C) 2009 Acoustical Society of America. [DOI: 10.1121/1.3117377]
Experimental modeling and design optimization of push-pull electret loudspeakers
(2010-04-01)
Bai, Mingsian R.
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Wang, Chun-Jen
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Chiang, Dar-Ming
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Lin, Shu-Ru
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機械工程學系
A fully experimental modeling technique and a design optimization procedure are presented in this paper for push-pull electret loudspeakers. Conventional electrical impedance-based parameter identification methods are not completely applicable to electret speakers due to the extremely weak electromechanical coupling. This prompts the development of an experimental technique for identifying the electroacoustic parameters of the electret speakers. Mechanical parameters are identified from the membrane velocity measured using a laser vibrometer. The voltage-force conversion factor and the motional impedance are estimated, with the aid of a test-box method. This experimentally identified model serves as the simulation platform for predicting the response of the electret loudspeaker and optimizing the design. Optimal parameters are calculated by using the simulated annealing (SA) algorithm to fulfill various design goals and constraints. Either the comprehensive search for various parameters or the simple search for the optimal gap distance can be conducted by this SA procedure. The results reveal that the optimized design has effectively enhanced the performance of the electret loudspeaker.
Expert diagnostic system for moving-coil loudspeakers using nonlinear modeling
(2009-02-01)
Bai, Mingsian R.
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Huang, Chau-Min
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機械工程學系
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Department of Mechanical Engineering
This work aims at the development of an expert diagnostic system for moving-coil loudspeakers. Special emphasis is placed on the defects resulting from loudspeaker nonlinearities. As a loudspeaker operates in the large signal domain, nonlinear distortions may arise and impair sound quality. Analysis of nonlinear responses can shed light on potential design faults of a loudspeaker. By exploiting this fact, this expert diagnostic system enables classification of design faults using a defect database alongside an intelligent fault inference module. Six types of defects are investigated in this paper. A large signal model based on electromechanical analogous circuits is employed for generating the defect database, through which a neural-fuzzy network is utilized for inferring the defect types. Numerical simulations and experimental investigations were undertaken for validating the loudspeaker diagnostic system. (C) 2009 Acoustical Society of America. [DOI: 10.1121/1.3058639]
Implementation issues of the nearfield equivalent source imaging microphone array
(2011-01-31)
Bai, Mingsian R.
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Lin, Jia-Hong
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Tseng, Chih-Wen
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機械工程學系
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Department of Mechanical Engineering
This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources. (C) 2010 Elsevier Ltd. All rights reserved.
Integration of a quantitative feedback theory (OFT)-based active noise canceller and 3D audio processor to headsets
(2007-10-01)
Bai, Mingsian R.
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Lin, Jianliang
;
機械工程學系
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Department of Mechanical Engineering
This paper seeks to enhance the quality of spatial sound reproduction by integrating two advanced signal processing technologies, active noise control (ANC) and three-dimensional (3D) audio, to a headset. The ANC module of the headset is designed based on the quantitative feedback theory (QFT), which is a unified theory that emphasizes the use of feedback for achieving the desired system performance tolerances in the face of plant uncertainties and plant disturbances. Performance, stability and robustness of the closed-loop system have been taken into account in the loop-shaping procedure within a general framework of the QFT On the other hand, 3D audio processing algorithms including the head-related-transfer-function and the reverberator are realized on the platform of a fixed-point digital signal processor Listening tests were conducted to evaluate the proposed system in terms of various subjective performance indices. The experimental results revealed that the 3D headset is capable of delivering superior rendering quality of localization and spaciousness, with the aid of the ANC module.
Intelligent classification system of sound signals and method thereof
(2008-04-03)
Bai, Mingsian R.
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Chen, Meng-Chun
A system that integrates various intelligent classification techniques and preprocessing algorithms is provided. A feature extracting unit receives audio signals and extracts audio features for identification by using various descriptors; a preprocessing unit normalized the data for data consistency; a classification unit classifying audio signals into several categories according to the audio features.
Method for parameter identification and parameter optimization of microspeakers
(2007-11-08)
Bai, Mingsian R.
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Chen, Rong Liang
The present invention discloses a method for parameter identification and parameter optimization of microspeakers. Measurement procedures for identifying electromechanical constants of microspeaker and a GUI are developed to facilitate estimation of electroacoustic parameters of the microspeaker under test. In light of the thus identified microspeaker parameters, a parameter optimization procedure is carried out to obtain the design that attains the best acoustic performance with minimum harmonic distortion.
Method for virtual bass synthesis
(2007-11-01)
Bai, Mingsian R.
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Tseng, Wen-Liang
;
Lin, Wan-Chi
This invention relates to a method for virtual bass synthesis. The low frequency signal is attained by applying a low pass filter to the original. In order to reduce the operations, process of down sampling the low frequency signal, moving the low frequency signal to a series of harmonics whose frequencies are integral times as large as the frequency of low frequency signals, and then up sampling them are provided. By means of psycho-acoustic theory, the weights of harmonics are attained and applied to the harmonics. Finally the weighted harmonics are combined to produce the bass signal. As the result, the virtual bass effect which is almost the same as the low frequency of the original audio signal can be accomplished. Because the harmonic signals are high frequency ones, the virtual effect can be made in the panel speakers or ordinary low-end speakers.
Method for virtual bass synthesis
(2011-04-21)
Bai, Mingsian R.
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Tseng, Wen-Liang
;
Lin, Wan-Chi
This invention relates to a method for virtual bass synthesis. The low frequency signal is attained by applying a low pass filter to the original. In order to reduce the operations, process of down sampling the low frequency signal, moving the low frequency signal to a series of harmonics whose frequencies are integral times as large as the frequency of low frequency signals, and then up sampling them are provided. By means of psycho-acoustic theory, the weights of harmonics are attained and applied to the harmonics. Finally the weighted harmonics are combined to produce the bass signal. As the result, the virtual bass effect which is almost the same as the low frequency of the original audio signal can be accomplished. Because the harmonic signals are high frequency ones, the virtual effect can be made in the panel speakers or ordinary low-end speakers.
Multirate synthesis of reverberators using subband filtering
(2009-04-10)
Bai, Mingsian R.
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Ou, Kwuen-Yieng
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Zeung, Pingshun
;
機械工程學系
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Department of Mechanical Engineering
As an important element in three-dimensional audio reproduction, reverberators serve to enhance the sense of spaciousness by synthesizing the sound reflections and reverberations pertaining to a particular listening environment. However, direct implementation of room responses generally proves impractical due to the extreme complexity of system dynamics. To overcome the difficulties encountered in long convolutions, an efficient multirate signal processing technique is proposed in this paper on the basis of subband filtering scheme. In this method, the frequency dependent property of reverberation time is incorporated into the design procedure. Further simplification follows from the synthesized infinite impulse response (IIR) implementation for each subband. Specifically, the regular IIR filter, the cascaded comb filters and nested allpass filters, and the cascaded comb filter and an FIR filter are developed to implement the subband filters. Genetic algorithms (GA) are employed to optimize the parameters of reverberators. Subjective listening tests are carried out to assess the performance of the proposed techniques. The results demonstrate that the implemented multirate reverberators are capable of delivering natural sensation of room reverberations, as compared to direct measurement. (C) 2008 Elsevier Ltd. All rights reserved.