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Research And Implementation On Adaptive Active Reduction Earmuffs Based On DSP

Posted on:2020-11-07Degree:MasterType:Thesis
Country:ChinaCandidate:Y R HuangFull Text:PDF
GTID:2392330620962278Subject:Information and Communication Engineering
Abstract/Summary:PDF Full Text Request
Active noise control technology is based on the principle of acoustic counteracting.With flexible structure and good control effect of low frequency noise,it has gradually become a research hotspot in the field of noise control.The research object of this paper is the noise in military battle tank cabin.The high intensity,the long duration and the concentrated frequency of cabin noise is existed in the low frequency band.The crew will be exposed to this noise environment for a long time,which will cause tremendous harm of heart,so it is of great practical significance to study the noise protection of crew in tank cabin.As a successful example of the application of active noise control technology,adaptive and active noise reduction earmuffs are equipped with controllers on the basis of traditional passive earmuffs to further improve the noise reduction effect of complex low-frequency noise in cabin.The main research work of this paper is as follows:(1)Combining with the practical application scenarios and noise characteristics,the principle,structure and characteristics of various active noise reduction systems are studied,including digital and analog,feedforward and feedback,single channel and multi-channel.The existing adaptive noise reduction algorithms and digital filter structures are compared,and a digital composite single channel active noise reduction earmuff structure is determined.(2)Least Mean Square(LMS)algorithm is widely used in active noise reduction due to its advantages of small computation and hardware implementation.In order to solve the contradiction between convergence speed and stability in LMS algorithm,a new variable step size LMS algorithm is proposed.Feedback mechanism is introduced into the algorithm.The step size factor update is associated with the error signal of the last iteration,and the anti-interference ability of the algorithm is improved.The new updating model solves the problem that the step factor changes too fast.The parameters of the model are adjusted and determined by experiments.Compared with the algorithms in the same field in recent three years,the simulation results show that the performance of the algorithm improves in convergence speed,steady-state error and so on.(3)Aiming at the problem of time delay in digital composite structure,the causes and solutions of system delay caused by secondary channel are studied.Secondary channel compensation is introduced to solve the problem of time delay.Secondary channel transfer function needs to be obtained.Based on the existing methods,an improved secondary channel identification model is proposed.The model uses additional Gauss white noise and normalized LMS algorithm to identify the secondary channel.The channel is identified offline,and the influencing factors of the model are determined by experiments.Finally,the importance of secondary channel compensation and the effectiveness of the improved identification model are proved by simulation and comparison.(4)Aiming at the complex noise environment in tank cabin,the standing wave sound field formed in enclosed three-dimensional space and the diffraction sound field formed in active noise reduction earmuffs are studied,and the layout scheme of secondary sound source and error microphone is determined.For the most difficult impulsive weapon noise to be cancelled in noise environment,a steady-state distributed noise model is established,the existing solutions are studied,and a nonlinear compression algorithm based on arc-tangent transform is proposed.The algorithm uses arc-tangent function to transform fractional moments of high impulsive noise,and introduces characteristic coefficient as compression factor to filter the compressed noise when the impulsive noise is too strong.The simulation results and comparison show that the algorithm can track and cancel the impulse noise very well.(5)The hardware and software system based on digital signal processing(DSP)chip TMS320VC5509 A is designed and implemented.The hardware system includes minimum DSP system,audio processing circuit,external memory expansion circuit,etc.The software system is based on code compser Studio(CCS)of the series of TMS320,including hardware module configuration and noise reduction program.Finally,single-frequency noise,white noise and actual tank cabin noise are used to verify the noise reduction ability of the system.
Keywords/Search Tags:active noise reduction system, variable step size LMS algorithm, secondary channel modeling, impulse noise, DSP system
PDF Full Text Request
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