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Speech Separation Algorithm Using Gated Recurrent Network Based on Microphone Array
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作者 Xiaoyan Zhao Lin Zhou +2 位作者 Yue Xie Ying Tong Jingang Shi 《Intelligent Automation & Soft Computing》 SCIE 2023年第6期3087-3100,共14页
Speech separation is an active research topic that plays an important role in numerous applications,such as speaker recognition,hearing pros-thesis,and autonomous robots.Many algorithms have been put forward to improv... Speech separation is an active research topic that plays an important role in numerous applications,such as speaker recognition,hearing pros-thesis,and autonomous robots.Many algorithms have been put forward to improve separation performance.However,speech separation in reverberant noisy environment is still a challenging task.To address this,a novel speech separation algorithm using gate recurrent unit(GRU)network based on microphone array has been proposed in this paper.The main aim of the proposed algorithm is to improve the separation performance and reduce the computational cost.The proposed algorithm extracts the sub-band steered response power-phase transform(SRP-PHAT)weighted by gammatone filter as the speech separation feature due to its discriminative and robust spatial position in formation.Since the GRU net work has the advantage of processing time series data with faster training speed and fewer training parameters,the GRU model is adopted to process the separation featuresof several sequential frames in the same sub-band to estimate the ideal Ratio Masking(IRM).The proposed algorithm decomposes the mixture signals into time-frequency(TF)units using gammatone filter bank in the frequency domain,and the target speech is reconstructed in the frequency domain by masking the mixture signal according to the estimated IRM.The operations of decomposing the mixture signal and reconstructing the target signal are completed in the frequency domain which can reduce the total computational cost.Experimental results demonstrate that the proposed algorithm realizes omnidirectional speech sep-aration in noisy and reverberant environments,provides good performance in terms of speech quality and intelligibility,and has the generalization capacity to reverberate. 展开更多
关键词 Microphone array speech separation gate recurrent unit network gammatone sub-band steered response power-phase transform spatial spectrum
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Sound Source Localization Based on SRP-PHAT Spatial Spectrum and Deep Neural Network 被引量:3
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作者 Xiaoyan Zhao Shuwen Chen +1 位作者 Lin Zhou Ying Chen 《Computers, Materials & Continua》 SCIE EI 2020年第7期253-271,共19页
Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transf... Microphone array-based sound source localization(SSL)is a challenging task in adverse acoustic scenarios.To address this,a novel SSL algorithm based on deep neural network(DNN)using steered response power-phase transform(SRP-PHAT)spatial spectrum as input feature is presented in this paper.Since the SRP-PHAT spatial power spectrum contains spatial location information,it is adopted as the input feature for sound source localization.DNN is exploited to extract the efficient location information from SRP-PHAT spatial power spectrum due to its advantage on extracting high-level features.SRP-PHAT at each steering position within a frame is arranged into a vector,which is treated as DNN input.A DNN model which can map the SRP-PHAT spatial spectrum to the azimuth of sound source is learned from the training signals.The azimuth of sound source is estimated through trained DNN model from the testing signals.Experiment results demonstrate that the proposed algorithm significantly improves localization performance whether the training and testing condition setup are the same or not,and is more robust to noise and reverberation. 展开更多
关键词 Sound source localization microphone array steered response power-phase transform(srp-phat)spatial spectrum deep neural network
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一种基于离散时延的鲁棒声源三维定位方法 被引量:2
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作者 蔡卫平 吴镇扬 《东南大学学报(自然科学版)》 EI CAS CSCD 北大核心 2009年第1期1-5,共5页
为了减少相位变换加权的可控响应功率(SRP-PHAT)声源定位算法的计算量,提出一种基于离散时延的改进算法.该方法首先利用FFT将麦克风阵列的每一帧接受信号变换到频域,然后在频域补零至16倍帧长,再运用IFFT将所有麦克风对的广义互相关函... 为了减少相位变换加权的可控响应功率(SRP-PHAT)声源定位算法的计算量,提出一种基于离散时延的改进算法.该方法首先利用FFT将麦克风阵列的每一帧接受信号变换到频域,然后在频域补零至16倍帧长,再运用IFFT将所有麦克风对的广义互相关函数在搜索之前计算好,从而可大幅度减少计算量.频域补零提高了广义互相关函数的采样率,因而由时延离散带来的定位误差很小.仿真结果表明,无论在远场还是近场条件下,该算法均能将计算量降低一个数量级而保持原算法的鲁棒性. 展开更多
关键词 麦克风阵列 声源定位 srp-phat算法
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基于混沌人工蜂群优化的声源定位算法
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作者 郭业才 马伟伟 朱赛男 《系统仿真学报》 CAS CSCD 北大核心 2017年第6期1168-1173,共6页
针对传统可控响应功率和相位变换(SRP-PHAT)的声源定位算法计算量大、实时性差等问题,提出了一种基于混沌人工蜂群优化的SRP-PHAT声源定位算法。该算法将阵列的波束输出功率作为目标代价函数,建立声源定位数学模型,利用混沌的全局覆盖... 针对传统可控响应功率和相位变换(SRP-PHAT)的声源定位算法计算量大、实时性差等问题,提出了一种基于混沌人工蜂群优化的SRP-PHAT声源定位算法。该算法将阵列的波束输出功率作为目标代价函数,建立声源定位数学模型,利用混沌的全局覆盖性和人工蜂群算法的收敛速度快、鲁棒性强等特点,实现了声源的三维精确定位。实验结果表明,该算法计算量小、定位精度高,能满足实际系统的定位需求。 展开更多
关键词 混沌映射 人工蜂群 可控响应功率和相位变换 声源定位算法
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