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论文中文题名:

 基于麦克风阵列的语音增强方法研究    

姓名:

 吴宝桐    

学号:

 19307205010    

保密级别:

 公开    

论文语种:

 chi    

学科代码:

 085208    

学科名称:

 工学 - 工程 - 电子与通信工程    

学生类型:

 硕士    

学位级别:

 工程硕士    

学位年度:

 2022    

培养单位:

 西安科技大学    

院系:

 通信与信息工程学院    

专业:

 电子与通信工程    

研究方向:

 语音增强    

第一导师姓名:

 贺顺    

第一导师单位:

 西安科技大学    

论文提交日期:

 2023-01-05    

论文答辩日期:

 2022-12-07    

论文外文题名:

 Research on speech enhancement method based on microphone array    

论文中文关键词:

 波束形成技术 ; 麦克风阵列 ; 语音增强 ; 广义旁瓣抵消器    

论文外文关键词:

 Beamforming technology ; Microphone array ; Voice enhancement ; Generalized sidelobe canceller    

论文中文摘要:

在实际应用中,语音通信过程中总会受到外界噪声的影响,使用语音增强方法能够将原始语音中混有的噪声消除,就显得非常重要了。使用多个麦克风搭建麦克风阵列模型,具有很强的空间选择性,同时能够获取多个声源信号,能够定位、自动检测以及在一定范围内可以跟踪说话人,对接收到的信号能抑制环境噪声的干扰,效果非常明显。然而在实际应用中,环境中的噪声是复杂多变的,针对复杂多变的声学环境问题,研究基于麦克风阵列的语音增强方法,非常有实际应用价值。本文在分析麦克风阵列语音信号处理理论基础上研究了几种传统麦克风阵列中语音增强算法,并提出两种改进算法,具体工作如下:

(1)广义旁瓣抵消器(GSC)算法能够有效抑制相干噪声,但对非相干噪声抑制能力不强,并且GSC结构中阻塞矩阵不能完全抑制目标语音信号,导致语音泄露。针对上述问题,本文提出基于改进谱减法的GSC语音增强算法(ASS-GSC),该算法能对GSC结构中阻塞矩阵(BM)的方向参数进行自适应的调整,能降低语音信号在BM模块中的泄露,再将GSC处理完的信号输入改进的谱减法中,通过对过减因子和增益补偿因子的调节,消除残留的音乐噪声。仿真结果表明,所提ASS-GSC算法对噪声谱的估算比较准确,不仅降低了混有的音乐影响,同时有效抑制非相干噪声的干扰,相较于经典的RSS-GSC算法、GSC与谱减法相结合的方法,该算法输出的语音质量更高。

(2)GSC与后置滤波相结合的算法中,该算法上支路延时-求和波束形成算法(DBF),要求麦克风阵元的数量必须超过一定规模,并且该算法只能在较大信噪比情形下去除非相干噪声,针对这一问题,本文提出一种改进后置滤波方法的GSC算法,该算法引入多源选择算法(MSS)选取能力最大的一路语音信息,提高信噪比,同时解决FBF算法对阵元数量的要求,并实现语音信号中相干和非相干噪声的抑制,从而达到语音增强的目的。仿真结果表明,相较于卷积传递函数广义旁瓣抵消器算法(CTF-GSC)的多通道后置滤波方法、传输函数比率(TF-GSC)算法及其优化方法,所提改进算法对非相干噪声的消除能力更强,具有较强的鲁棒性。

论文外文摘要:

In practical applications, speech communication is always affected by external noise. It is very important to use speech enhancement methods to eliminate the noise mixed in the original speech. Using multiple microphones to build a microphone array model has strong spatial selectivity. At the same time, it can obtain multiple sound source signals. It can locate, automatically detect and track the speaker within a certain range. It can suppress the interference of environmental noise on the received signal, with obvious results. However, in practical applications, the noise in the environment is complex and changeable. For the complex and changeable acoustic environment, it is very valuable to study the speech enhancement method based on microphone array. Based on the analysis of microphone array speech signal processing theory, this paper studies several speech enhancement algorithms in traditional microphone arrays, and proposes two improved algorithms. The specific work is as follows:

(1) The generalized sidelobe canceller (GSC) algorithm can effectively suppress coherent noise, but it has weak ability to suppress incoherent noise, and the blocking matrix in the GSC structure can not completely suppress the target speech signal, resulting in speech leakage. To solve the above problems, this paper proposes a GSC speech enhancement algorithm based on improved spectral subtraction (ASS-GSC). This algorithm can adaptively adjust the direction parameters of the blocking matrix (BM) in the GSC structure, reduce the leakage of voice signals in the BM module, and then input the signal processed by GSC into the improved spectral subtraction to eliminate the residual music noise by adjusting the over subtraction factor and gain compensation factor. The simulation results show that the proposed ASS-GSC algorithm is more accurate in estimating the noise spectrum, which not only reduces the influence of mixed music, but also effectively suppresses the interference of incoherent noise. Compared with the classical RSS-GSC algorithm and the method of combining GSC with spectral subtraction, the output voice quality of this algorithm is higher.

(2)In the algorithm combining GSC and post filtering, the branch delay sum beamforming algorithm (DBF) on this algorithm requires that the number of microphone array elements must exceed a certain scale, and this algorithm can only be used in the case of large signal to noise ratio except for coherent noise. To solve this problem, this paper proposes a GSC algorithm that improves the post filtering method. This algorithm introduces the multi-source selection algorithm (MSS) to select the most powerful voice information, Improve the signal-to-noise ratio, solve the requirements of FBF algorithm on the number of elements, and achieve the suppression of coherent and incoherent noise in speech signals, so as to achieve the purpose of speech enhancement. The simulation results show that, compared with the multi-channel post filtering method, the transfer function ratio (TF-GSC) algorithm and its optimization method of the convolutional transfer function generalized sidelobe canceller algorithm (CTF-GSC), the improved algorithm has stronger ability to eliminate incoherent noise and stronger robustness.

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中图分类号:

 TN912.35    

开放日期:

 2023-04-12    

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