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基于深度学习的混响感知麦克风阵列语音增强

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[目的]针对基于深度神经网络频谱估计的麦克风阵列算法存在数据依赖的问题,提出了一种基于深度学习的混响感知麦克风阵列语音增强算法.[方法]首先利用麦克风阵列波束形成输出与原始信号做互相关,以近似房间冲激响应的形式获取当前环境的混响特性作为LSTM网络的输入,网络模型以干净语音为目标进行训练从而输出房间冲激响应泛化向量,最后通过组合近似房间冲激响应与房间冲激响应泛化向量获得后置抗混响滤波器系数,实现语音增强.[结果]仿真和实验结果中,与波束形成、加权预测误差算法和传统深度学习去混响算法相比,所提出的方法在不同混响场景下具有更好的表现.[结论]本文方法在不同混响场景下都具有相对稳定的抗混响能力,具有较好的泛化性能.
Reverberation-aware microphone array speech-enhancement algorithm based on deep-learning
[Objective]The technique of microphone array has been extensively applied for enhancing speech by means of the exploration of spatial information provided by multiple microphone channel.However,due to diverse reverberation characteristics produced by different sizes,different boundary materials and different reflectors,the speech enhance performance of microphone array are deteriorated significantly.In recent years,the deep-learning optimized microphone array signal processing has been investigated to remedy the problem caused by reverberation,which endures the data dependence and thus cannot adapt to the reverberation scene that is excluded from the training data.In this paper,a novel reverberation-aware(RA)microphone array speech enhancement algorithm is proposed to first obtain the reverberant feature and then design a deep-learning model to decouple the negative impact of environments,thus facilitating environment adaptive microphone array speech enhancement under diverse reverberant scenarios.[Methods]The proposed RA microphone array speech enhancement algorithm consists of training stage and testing stage.Specifically,in the training stage,the simulated reverberant signal is used for obtaining approximate room impulse response(ARIR)by correlating the reverberant signal with its beamforming output.Then,with the clean speech as training target,a RA model is designed by adopting ARIR and the beamformed signal as the training input.Consequently,a diverse room impulse response(RIR)generalized vector(RGV)to generalize the de-reverberation model with respect to RIR as well as the uncontrolled speech can be produced.In the practical testing stage,the practical ARIR is similarly obtained by correlating the received reverberant signal with its beamforming output.Afterward the resulting RGV is used to convolve with the practical ARIR to obtain the coefficients of a post de-reverberation filter,which exerts to remove the reverberation corresponding to ARIR.[Results]Performance of the proposed RA speech enhancement algorithm is quantitatively evaluated through simulations and experiments,in which the classic filter and sum beamforming(FSB)algorithm,weighted prediction error(WPE)algorithm,and DNN-WPE algorithm are chosen as comparative methods.The perceptual evaluation of speech quality(PESQ)scores and the speech-to-reverberation modulation energy ratio(SRMR)serve as evaluation metrics for assessing speech quality.Also,the THCHS-30 dataset is utilized for training and testing.In the case of environment match,those datasets for model training and testing originate from the same room,whereas,in the case of environment mismatch,those datasets for model training and testing originate from different rooms.In the simulation,artificial RIR with different reverberation levels are constructed based on the IMAGE toolbox,and speech signals with different reverberation levels can be generated by convolving the aforementioned original pure corpus with the artificial RIR to simulate the reverberant multichannel received signals of microphone arrays.Simulation results show that,under the condition of environment match,both DNN-WPE and the proposed RA deep learning algorithms outperform the traditional FSB algorithm and WPE algorithm at all reverberation levels.However,in case of environment mismatch,the performance of both DNN-WPE and the proposed RA algorithms worsen.Notably,while the DNN-WPE experiences significant performance degradation in terms of PESQ and SRMR,the proposed RA algorithm continues to exhibit better performance than the traditional FSB algorithm and WPE algorithm do.In the practical experiment,speech data is recorded with reverberation times of 0.25,0.4,and 0.6 s in a reverberation laboratory with adjustable reverberant level.Experimental results reveal that,by comparing the environment mismatch case to the environment match case,the DNN-WPE algorithm demonstrates significant performance degradation,whereas the proposed RA algorithm exhibits much more stability in terms of the PESQ and SRMR.This trend indicates that the proposed RA algorithm outperforms the DNN-WPE method in terms of environmental tolerance,consistently resembling results of the simulation.[Conclusions]Based on the evaluation and comparison results of different algorithms obtained via simulations and practical experiments,the RA microphone array speech-enhancement algorithm proposed in this paper is capable of achieving a satisfactory performance under diverse reverberation environments.In the RA algorithm,ARIR is used as an input to the model,thus somewhat reducing the dependence of the neural network on training data.In future research,we will consider the combination of other models and training methods to explore the potential of ARIR in improving the generalization ability of the model.

reverberationmicrophone arraybeamformingroom impulse responsedeep learningLSTM

何伟、刘雨佶、童峰、康元勋、冯万健

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厦门大学海洋与地球学院,福建厦门 361005

导航与位置服务技术国家地方联合工程研究中心(厦门大学),福建厦门 361005

厦门亿联网络技术股份有限公司,福建厦门 361015

混响 麦克风阵列 波束形成 房间冲激响应 深度学习 长短时记忆

上海市科委科技创新行动计划厦门市海洋产业项目

21DZ120550222CZB012HJ13

2024

厦门大学学报(自然科学版)
厦门大学

厦门大学学报(自然科学版)

CSTPCD北大核心
影响因子:0.449
ISSN:0438-0479
年,卷(期):2024.63(2)
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