[1]梁瑞宇,赵力,王青云,等.实时多通道数字助听器降噪算法[J].东南大学学报(自然科学版),2016,46(1):13-17.[doi:10.3969/j.issn.1001-0505.2016.01.003]
 Liang Ruiyu,Zhao Li,Wang Qingyun,et al.Real-time noise reduction algorithm for multi-channel digital hearing aids[J].Journal of Southeast University (Natural Science Edition),2016,46(1):13-17.[doi:10.3969/j.issn.1001-0505.2016.01.003]
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实时多通道数字助听器降噪算法()
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《东南大学学报(自然科学版)》[ISSN:1001-0505/CN:32-1178/N]

卷:
46
期数:
2016年第1期
页码:
13-17
栏目:
信息与通信工程
出版日期:
2016-01-20

文章信息/Info

Title:
Real-time noise reduction algorithm for multi-channel digital hearing aids
作者:
梁瑞宇12赵力1王青云12邹采荣13荆丽1
1东南大学信息科学与工程学院, 南京 210096; 2南京工程学院通信工程学院, 南京 211167; 3广州大学机械与电气工程学院, 广州 510006
Author(s):
Liang Ruiyu12 Zhao Li1 Wang Qingyun12 Zou Cairong13 Jing Li1
1School of Information Science and Engineering, Southeast University, Nanjing 210096, China
2School of Communication Engineering, Nanjing Institute of Technology, Nanjing 211167, China
3School of Mechanical and Electric Engineering, Guangzhou University, Guangzhou 510006, China
关键词:
降噪 多通道助听器 自适应维纳滤波 声压级
Keywords:
noise reduction multi-channel hearing aid adaptive Wiener filtering sound pressure level
分类号:
TN912.3
DOI:
10.3969/j.issn.1001-0505.2016.01.003
摘要:
在兼顾降噪性能和功耗的基础上,提出了一种实时多通道数字助听器降噪算法.首先,将输入信号分解为16个子带,计算每个子带的声压级,并基于估计的声压级来计算子带噪声和语音概率;然后,利用直接判决方法计算子带信号的先验信噪比和后验信噪比;最后,计算子带增益函数以实现自适应降噪.将该算法与改进谱减法、自适应维纳滤波法和调制深度法进行了比较.结果表明:与其他3种算法相比,在10 dB白噪声的情况下,本文算法输出的平均信噪比减少约3 dB,主观语音质量评估得分最多提高0.90;在4种噪声环境下其平均主观语音质量评估得分提高0.41;所提算法采用子带声压级计算取代信号功率谱估计,节省了快速傅里叶变换的计算量,其时延较其他3种算法至少降低50%.
Abstract:
A real-time noise reduction algorithm for multi-channel digital hearing aids is proposed based on the balance between noise reduction performance and power consumption. First, the input signal is decomposed into 16 subbands and the sound pressure level(SPL)of each subband is calculated. Based on the estimated SPL, the subband noise and the speech probability are computed. Then, the priori signal noise ratio(SNR)and the posteriori SNR of the subband signal are calculated by the direct decision method. Finally, the gain function is calculated to adaptively reduce noises. And the proposed algorithm is compared with the improved spectral subtraction, adaptive Wiener filter and the algorithm based on the modulation depth. The experimental results show that compared with the other three algorithms, the average SNR of the proposed algorithm decreases by about 3 dB and the perceptual evaluation of speech quality(PESQ)is at most improved by 0.90 when the SNR of the white noise is 10 dB. In addition, the average output PESQ is improved by 0.41 in four kinds of noisy environments. In the proposed algorithm, the estimation of the power spectrum is replaced by the calculation of the subband SPL and the fast Fourier transform computation is reduced, inducing at least 50% decrease of the time-delay compared with the other three algorithms.

参考文献/References:

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备注/Memo

备注/Memo:
收稿日期: 2015-08-30.
作者简介: 梁瑞宇(1978—),男,博士,副教授;赵力(联系人),男,博士,教授,博士生导师,zhaoli@seu.edu.cn.
基金项目: 国家自然科学基金资助项目(61273266,61301219,61375028)、江苏省自然科学基金资助项目(BK20130241).
引用本文: 梁瑞宇,赵力,王青云,等.实时多通道数字助听器降噪算法[J].东南大学学报(自然科学版),2016,46(1):13-17. DOI:10.3969/j.issn.1001-0505.2016.01.003.
更新日期/Last Update: 2016-01-20