一种近场环境下的麦克风阵列语音增强方法

针对语音源位于阵列近场而干扰噪声源位于阵列远场的声学环境,本文提出了一种基于近场双自适应波束形成的麦克风阵列语音增强方法。该方法利用近场声波波前的特点,主通道采用最小方差无失真响应准则的近场优化波束形成器,辅助通道采用双自适应波束形成技术,从而有效地抑制了混响和噪声对语音信号的影响。仿真实验结果表明,在房间混响条件下,本文方法具有良好的噪声抑制性能。     

基于传声器阵列语音增强方法研究

传声器阵列通过对拾取的多路语音信号进行分析与处理,能取得改进语音质量、消除背景噪声和提高语音可懂度等明显效果,现已成为语音信号增强的一个重要的研究领域。介绍了基于传声器阵列的自适应波束形成方法,该方法采用GSC结构基于TF-GSC的最优后置滤波算法。仿真实验结果表明,该自适应波束形成器对干扰有很好的消除作用,对阵元的增益误差、位置误差不敏感,可以取得较好的语音增强效果。     

面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法

二元麦克风小阵列在手机、助听器等受空间、成本以及运算能力限制的设备中被广泛研究用以提高目标语音质量。二元麦克风小阵列中语音增强算法主要包括波束形成方法以及相干性滤波器方法。波束形成方法的思想是利用目标声源相对阵列的位置关系获取相应的时域和空域信息,可以保留目标声源方向的信号而抑制其他方向的干扰信号;相干性滤波器方法则通过阵元间不同信号的相关性进行噪音抑制。考虑这两种类型方法的优点,本文提出一种面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法,通过在广义旁瓣抵消器的固定波束形成支路上使用相干性滤波器,提高固定波束形成输出信号的信噪比,然后在广义旁瓣抵消器自适应支路利用阵列的时域和空域信息对固定波束形成支路输出的信号中残余噪音进行估计,进而获得增强后目标输出信号。仿真和实际试验表明,本文提出的算法明显优于单独使用小阵列波束形成算法和相干性滤波器算法。      

GSVD-based optimal filtering for single and multimicrophone speech enhancement

A generalized singular value decomposition (GSVD) based algorithm is proposed for enhancing multimicrophone speech signals degraded by additive colored noise. This GSVD-based multimicrophone algorithm can be considered to be an extension of the single-microphone signal subspace algorithms for enhancing noisy speech signals and amounts to a specific optimal filtering problem when the desired response signal cannot be observed. The optimal filter can be written as a function of the generalized singular vectors and singular values of a speech and noise data matrix. A number of symmetry properties are derived for the single-microphone and multimicrophone optimal filter, which are valid for the white noise case as well as for the colored noise case. In addition, the averaging step of some single-microphone signal subspace algorithms is examined, leading to the conclusion that this averaging operation is unnecessary and even suboptimal. For simple situations, where we consider localized sources and no multipath propagation, the GSVD-based optimal filtering technique exhibits the spatial directivity pattern of a beamformer. When comparing the noise reduction performance for realistic situations, simulations show that the GSVD-based optimal filtering technique has a better performance than standard fixed and adaptive beamforming techniques for all reverberation times and that it is more robust to deviations from the nominal situation, as, e.g., encountered in uncalibrated microphone arrays.     

Low-bandwidth binaural beamforming

An efficient beamforming scheme for wireless binaural hearing aids is proposed that provides a trade-off between the transmission bit rate and the amount of noise reduction. It is proposed to transmit only the lowfrequency part of the signal from one hearing aid to the other, which is used in a binaural beamformer to generate the low-frequency part of the output. The high-frequency part is generated by a monaural beamformer using only the locally available microphone signals. The trade-off can be attained by adjusting the cutoff frequency of the lowpass filter. For speech sources with a 8 kHz bandwidth in the presence of an interfering source, it is shown that good performance can be achieved with a cutoff frequency of 4 kHz.     

Application of near-field optimum microphone arrays to hands-free mobile telephony

This paper discusses the application of fixed microphone arrays to speech pickup in mobile telephone applications. Array optimization techniques are used to design two broad-band beamformers for speech pickup in the near field. The first beamformer provides optimum gain for spatially incoherent noise while the second beamformer provides optimum gain in spherically isotropic noise. Array performance was measured using vehicular noise recorded under realistic driving conditions. Results obtained are in agreement with theoretical predictions for a spherically isotropic noise field and are comparable to previously reported results obtained using adaptive beamforming algorithms.     

谱相减与波束形成相结合的语音增强算法

根据阵列信号语音增强的思想,提出一种基于频城处理的谱相减与波束形成相结合的语音增强结构。结构为多路信号输入,每路含噪信号在谱相减后,增加了波束形成结构,不仅有效地消除了背景噪声,也抑制了谱相减后的音乐噪声。并使用该算法对实际环境中采集到的含噪语音信号进行了仿真,结果显示经过该系统处理后的增强语音的信噪比有了较大的提高,主观试听效果也很好。     

基于传声器阵列超增益波束形成的语音增强--频域方法

提出了一种在频域实现的基于传声器阵列超增益波束形成的语音增强方法。该方法利用小孔径线列阵端射方向具有超增益的特性,针对均匀噪声场,设计出相应的超增益权,形成超增益波束。基于超增益波束形成的输出相对常规处理,可大幅度提高信噪比。仿真了间距为0.05m的5元均匀线性传声器阵列接收到的端射方向带噪线性调频信号和语音信号,并进行超增益处理,获得12dB左右的阵增益,从而表明超增益传声器阵列具有优越的性能。     

基于传声器阵的两级自适应滤波语音增强系统

基于自适应噪声对消思想，提出一种基于传声器阵的自适应语音增强结构，该结构经过两级自适应滤波，分离出增强语音信号。计算机仿真结果显示该系统能有效去除背景噪声。     

语音增强及其消噪能力研究

语音增强技术可极大提高信噪比，解决由于环境噪声引起的语音通讯和识别性能下降的问题。目前常用的语音增强算法有频谱相减法，维纳滤波法，自适应抵消法等。文章提出一种将指向性麦克风和自适应抵消法相结合的方法，在仿真试验中取得了较好的结果。     

基于阵列抗串扰自适应噪声抵消的语音增强

本文提出了一种针对阵列交叉串扰信号的自适应噪声抵消方法,并将其用于麦克风阵语音增强.该方法仅使用两级滤波系统,计算量小,稳定性好,且对麦克风阵的几何结构及噪声类型均没有严格限制.试验表明,该方法消噪量大,对语音损伤小,语音增强效果显著,适用于多种噪声环境并易于实时实现.     

Output SNR analysis of integrated active noise control and noise reduction in hearing aids under a single speech source scenario

This paper analyses the output signal-to-noise ratio for a standard noise reduction scheme based on the multichannel Wiener filter and for an integrated active noise control and noise reduction scheme based on the filtered-X multichannel Wiener filter, both applied in a hearing aid framework that includes the effects of signal leakage through an open fitting and secondary path effects. In previous work, integrating noise reduction and active noise control has been shown to allow to compensate for effects of signal leakage and secondary path effects. These experimental results are now verified theoretically. The output signal-to-noise ratios are derived under a single speech source scenario. Theoretical results are then compared to simulations for a single noise source scenario and a multiple noise sources scenario.     

Adaptive multiresolution speech enhancement algorithm based on wavelet transform

In this paper, an adaptive multiresolution speech enhancement algorithm based on wavelet transform is put forward. It can make adaptive filtering to noise speech both at scales and among scales. So that the noise parts during the frequency intervals which decrease hearing quality mostly are reduced efficiently. Both the SNR and subject hearing quality of denoised speech are high and good.     

基于小波变换的自适应多分辨率语音增强算法

本文提出了一种基于小波变换的自适应多分辨率语音增强算法,它在尺度上和尺度间同时对受噪声污染的语音信号作自适应滤波处理,从而使得对听觉影响最严重的频段上的噪声被有效地滤除掉,滤波后的语音信噪比和主观听觉质量都得到了很大的改善。     

基于盲源分离理论的麦克风阵列信号有音/无音检测方法

该文提出一种在方向性噪声场中多路麦克风信号同时进行有音/无音检测(VAD)的方法。在方向性噪声场中,由于各个麦克风接收信号中的噪声彼此之间相关,因而,可以利用盲源分离理论将方向噪声与语音源信号分离,从而获得相对比较纯净的语音源信号。对分离出的语音源信号进行有音/无音检测,获得VAD结果,同时估计出各个麦克风信号相对于该信号的时延值。以相对纯净语音源信号的VAD检测结果为参考,将其分别平移相应的时延值,即可同时获得多路麦克风信号的VAD结果。计算机模拟结果表明,在方向性噪声场的多种情况下,该方法对具有加性噪声的多路麦克风信号均具有较好的有音/无音检测能力。     

Two‐Microphone Binary Mask Speech Enhancement in Diffuse and Directional Noise Fields

Two‐microphone binary mask speech enhancement (2mBMSE) has been of particular interest in recent literature and has shown promising results. Current 2mBMSE systems rely on spatial cues of speech and noise sources. Although these cues are helpful for directional noise sources, they lose their efficiency in diffuse noise fields. We propose a new system that is effective in both directional and diffuse noise conditions. The system exploits two features. The first determines whether a given time–frequency (T‐F) unit of the input spectrum is dominated by a diffuse or directional source. A diffuse signal is certainly a noise signal, but a directional signal could correspond to a noise or speech source. The second feature discriminates between T‐F units dominated by speech or directional noise signals. Speech enhancement is performed using a binary mask, calculated based on the proposed features. In both directional and diffuse noise fields, the proposed system segregates speech T‐F units with hit rates above 85%. It outperforms previous solutions in terms of signal‐to‐noise ratio and perceptual evaluation of speech quality improvement, especially in diffuse noise conditions.     

基于感知加权滤波的卡尔曼滤波语音增强

结合人耳听觉掩蔽效应,提出一种基于听觉感知加权的卡尔曼滤波语音增强方法。由于人耳对语音的感知主要是通过语音信号频谱分量幅度获得的,引入听觉感知加权滤波器在频域上使共振峰区域残留噪声更多,而共振峰之间及语音幅度谱较低的区域残留噪声减少,这样符合人耳的听觉特性,从而使得主观感觉到的噪声最小。采用语音质量感知评估对语音增强的效果进行评测,与传统的卡尔曼滤波语音增强算法相比,实验结果显示该算法提高了增强语音的质量。     

基于多任务疏表达的二元麦克风小阵列语音增强算法简

针对常规二元麦克风小阵列话音增强算法通常需要话音活动检测技术支持,并且难以有效抑制第一帧含目标信号的噪声。提出了一种基于多任务稀疏表达的二元麦克风小阵列话音增强算法,首先利用字典学习方法分别获得目标信号和噪声信号的过完备字典,然后利用 混合范数对信号在其字典上的表示系数进行正则化稀疏约束,使得2个阵元接收到信号中的噪声信号被抑制,而话音信号尽量保持不变,从而达到话音增强的目标。仿真和实验数据表明,无论开始位置是否含有目标话音信号,所提出的非话音活动检测支持的二元麦克风小阵列话音增强算法均能有效实现话音增强的目标。     

通信对抗中的自适应滤波技术

自适应滤波是在维纳滤波和Kalman滤波等线性滤波基础上发展起来的一种最佳滤波方法,具有较强的适应性和较优的滤波性能。这里将自适应滤波技术应用于电子对抗领域,利用自适应滤波原理计算出一个数字滤波器,对各信道的增益失配与相移失配进行精确的通道均衡补偿;利用自适应滤波方法设计具有特定频率响应的FIR滤波器,可实现时域宽带波束形成技术,并实现了基于自适应滤波的同平台干扰抵消技术。     

Robust Voice Activity Detection Based on Concept of Modulation Transfer Function in Noisy Reverberant Environments

Voice activity detection (VAD) is used to detect speech and non-speech periods from observed speech signals. It is an important front-end technique for many speech technology applications. Many VAD methods have been proposed. However most of them have been applied under clean or noisy conditions. Only a few methods have been proposed for reverberant conditions, particularly under noisy reverberant conditions. We therefore need to understand the ill effects of noise and reverberation on speech to design an accurate and robust method of VAD under noisy reverberant conditions. The ill effects of noise and reverberation for speech can be regarded as the modulation transfer function (MTF) under noisy and reverberant conditions. Therefore, our study is based on the MTF concept to reduce the ill effects of noise and reverberation on speech, and propose a robust VAD method that we obtained in this study. Noise reduction and dereverberation were first applied to the temporal power envelope of the speech signal to restore the temporal power envelope with this method. Then, power thresholding as a VAD decision was designed based on the restored temporal power envelope. A method of estimating the signal to noise ratio (SNR) was proposed to accurately estimate the SNR in the noise reduction stage. Experiments under both artificial and realistic noisy reverberant conditions were carried out to evaluate the performance of the proposed method of VAD and it was compared with conventional VAD methods. The results revealed that the proposed method significantly outperformed the conventional methods under artificial and realistic noisy reverberant conditions.