Design Of A Pickup System Based On Microphone Array

For a long time,extracting clean target speech from a noisy environment has been a subject of extensive research.Traditional single-microphone speech enhancement algorithms often lead to distortion of the target speech,and the sound quality cannot be guaranteed;while the microphone array technology can realize the localization and directional enhancement of the target sound source,and obtain obvious interference noise under the premise of ensuring that the target speech is not distorted Inhibitory effect.The microphone array-based sound source location and voice enhancement solution can not only be applied to audio and video conference calls to improve audio quality,but also can be used on smart voice terminals as the acoustic front-end processor of smart devices to obtain clearer voice commands,and then Improve the efficiency of human-computer interaction.In recent years,microphone array technology has been extensively studied and made considerable progress.This paper mainly studies the sound source localization algorithm and speech enhancement algorithm based on the microphone array,proposes a real-time sound source localization and speech enhancement system solution suitable for embedded devices,and builds a uniform circular microphone array based on the DSP chip.The hardware equipment of the sound pickup system is used as the landing platform of the scheme.The main work and innovations of this paper are as follows:(1)The background of microphone array technology and the development status at home and abroad are studied,especially the development history of sound source localization and speech enhancement technology based on microphone array.The basic knowledge of microphone array signal processing is studied.The generation method and characteristics of the voice signal are investigated;the preprocessing methods for the characteristics of the voice signal are learned and applied;the topological structure of the microphone array and the selection of the array element spacing that need to be considered when constructing the microphone array are studied.(2)A low-complexity sound source localization scheme based on improved correlation time delay estimation is proposed.Firstly,three classical microphone array sound sources are studied and analyzed: a sound source localization algorithm based on high-resolution spectrum estimation,a sound source localization algorithm based on controllable beam response technology,and a two-step sound source localization algorithm based on time difference of arrival(TDOA).Localization technology,and based on the sound source localization algorithm based on the time difference of arrival,research and propose a low-complexity sound source localization scheme based on improved correlation time delay estimation.The sound source localization scheme has three main innovations,namely: in the time delay estimation process,the weighting function of the generalized cross-correlation function is low-pass filtered to ensure the robustness of the time delay calculation in the noisy environment,while further reducing the algorithm calculation Complexity,to better ensure the real-time performance of the algorithm;when using the time delay estimation result to determine the sound source position,the global search strategy of the sound source localization algorithm based on the controllable beamforming technology is used for reference,and the minimum mean square error is used as the criterion to determine the sound Source orientation: For the dynamic frame number sound source location smoothing strategy under the scene of the dynamic change of the sound source orientation,the dynamic frame number smoothing strategy comprehensively guarantees the accuracy of the positioning result and the agility of tracking the sound source orientation change.Simulation experiments prove that the low-complexity sound source localization scheme based on improved correlation time delay estimation is better than ordinary generalized cross-correlation time delay estimation localization algorithms based on phase transformation.(3)A microphone array speech enhancement algorithm based on noise modeling and postfiltering is proposed.Firstly,several typical multi-channel speech enhancement algorithms are studied,including fixed beamforming,adaptive beamforming,and post-filtering algorithms.The implementation principles,advantages and disadvantages and applicable occasions of these typical algorithms are analyzed in detail.Based on the post-filtering algorithm,a microphone array speech enhancement algorithm based on noise modeling and post-filtering is proposed.For the consideration of real-time and robustness,the speech enhancement algorithm proposed in this paper uses a fixed beamformer and post-filter The form of cascade.The innovation of this algorithm lies in the comprehensive modeling and analysis of the target signal,directional interference signal and environmental noise in the sound field to obtain the optimal estimation of the target signal component.The simulation experiment proves that the performance of the microphone array speech enhancement algorithm based on noise modeling and post-filtering proposed in this paper is significantly improved compared to the traditional delay-sum beamforming algorithm.(4)The microphone array pickup system was designed and built,and the experiment proved the excellent performance of the microphone array pickup system.The sound pickup system is based on the embedded processor TMS320C6748,with TLV320AIC3106 low-power stereo audio codec and 2559 power amplifier;4 SPH0642HT5H-1 silicon microphones are used to construct a uniform circular microphone array.Based on this sound pickup system hardware equipment for experimental testing,the test results show that the actual performance of the microphone array pickup system designed in this paper is better.