基于能量损失率估计的传声器阵列语音增强

在延时相加波束形成和维纳滤波技术的基础上,提出了一种基于能量损失率估计的传声器阵列后滤波语音增强算法。该算法通过检测线性不等间距传声器阵列中各嵌套子阵在波束形成前后的能量变化来估计维纳滤波器的权系数,实现了语音增强的目标。在仿真数据集上的实验评估表明,相比原始语音,该算法增强后的语音在信噪比、对数谱距离和感知质量等指标上平均分别改善了17.1 dB,1.001和0.935,具有很好的应用前景。      

An iterative noise cross-PSD estimation for two-microphone speech enhancement

Among various speech enhancement methods, dual-microphone methods are of a great importance for their low cost implementation and for exploiting spatial-filtering benefits of the microphone arrays. Coherence based methods are well-known as efficient two-microphone noise reduction techniques. These techniques do not work well, when received noise signals are correlated. These can be improved when the cross power spectral density (CPSD) of noise is available. In this paper, we propose an iterative approach for estimation of the noise CPSD to be employed in coherence based methods. We compare the proposed iterative noise CPSD estimation with a noise CPSD estimation technique based on voice activity detector (VAD), both of which are employed in a two-microphone speech enhancement, separately. Evaluation results show that the two-microphone speech enhancement scheme utilizing the proposed noise CPSD estimation technique performs superior than the enhancement system using the VAD-based noise CPSD estimation.     

Kalman filter-based microphone array signal processing using the equivalent source model

This paper demonstrates that microphone array signal processing can be implemented by using adaptive model-based filtering approaches. Nearfield and farfield sound propagation models are formulated into state-space forms in light of the Equivalent Source Method (ESM). In the model, the unknown source amplitudes of the virtual sources are adaptively estimated by using Kalman filters (KFs). The nearfield array aimed at noise source identification is based on a Multiple-Input–Multiple-Output (MIMO) state-space model with minimal realization, whereas the farfield array technique aimed at speech quality enhancement is based on a Single-Input–Multiple-Output (SIMO) state-space model. Performance of the nearfield array is evaluated in terms of relative error of the velocity reconstructed on the actual source surface. Numerical simulations for the nearfield array were conducted with a baffled planar piston source. From the error metric, the proposed KF algorithm proved effective in identifying noise sources. Objective simulations and subjective experiments are undertaken to validate the proposed farfield arrays in comparison with two conventional methods. The results of objective tests indicated that the farfield arrays significantly enhanced the speech quality and word recognition rate. The results of subjective tests post-processed with the analysis of variance (ANOVA) and a post-hoc Fisher's least significant difference (LSD) test have shown great promise in the KF-based microphone array signal processing techniques.     

Single-channel speech enhancement method using reconstructive NMF with spectrotemporal speech presence probabilities

In this paper, a novel single microphone channel-based speech enhancement technique is presented. While most of the conventional nonnegative matrix factorization-based approaches focus on generating a basis matrix of speech and noise for enhancement, the proposed algorithm performs an additional process to reconstruct speech from noisy speech when these two elements are highly overlapped in selected spectral bands. This process involves a log-spectral amplitude based estimator, which provides the spectrotemporal speech presence probability to obtain a more accurate reconstruction. Moreover, the proposed algorithm applies an unsupervised learning method to the input noise, so it is adaptable to any type of environmental noise without a pre-trained dictionary. The experimental results demonstrate that the proposed algorithm obtains improved speech enhancement performance compared with conventional single channel-based approaches.     

Design and implementation of a MEMS microphone array system for real-time speech acquisition

Despite many attractive features and the potential for capturing sound in challenging acoustic environments, arrays with a large number of microphones have for a long time been discarded as a practical solution for speech acquisition. This is, among other reasons, due to the high production and computational costs. Only a few realizations of large microphone array systems have been documented, mainly for research and instrumentation use. The advent of MEMS microphones and computationally powerful off-the-shelf hardware has created new possibilities for microphone array development. We investigate a real life application, specifically the case of live sports broadcast, and the requirements that a such application imposes on a microphone array system. We present a system architecture of the first large (300 element circular array with a diameter of 2 m) MEMS microphone array system. In the proposed system, the latest technological advances are utilized to create a user-friendly array control interface. The array’s performance is examined in an anechoic chamber and on a crowded basketball field, and finally compared with existing solutions. The results illustrate the potential of a large MEMS microphone array as part of the technological development in sound acquisition for entertainment and security applications.     

Speech dereverberation method based on spectral subtraction and spectral line enhancement

Speech signals recorded with a distant microphone usually are interfered by the spatial reverberation in the room, which severely degrades the clarity and intelligibility of speech. A speech dereverberation method based on spectral subtraction and spectral line enhancement is proposed in this paper. Following the generalized statistical reverberation model, the power spectrum of late reverberation is estimated and removed from the reverberation speech by the spectral subtraction method. Then, according to the human auditory model, a spectral line enhancement technique based on adaptive post-filtering is adopted to further eliminate the reverberant components between adjacent speech formants. The proposed method can effectively suppress the spatial reverberation and improve the auditory perception of speech. The subjective and objective evaluation results reveal that the perceptual quality of speech is greatly improved by the proposed method.     

一种新的手机双麦克风消噪系统*

针对现有的手机双麦克风消噪系统无法应对多种复杂的噪声环境在消除噪声的同时会引起语音失真等问题,本文提出了一种新的手机双麦克风消噪系统,该系统将时域与频域处理相结合,在噪声估计和噪声消除两个方面均做了改进,结合双麦克风和单麦克风的噪声估计算法,提高了噪声估计的准确性,同时将基音检测与消噪处理相结合,在语音帧中估计语音基音频率,同时确定语音和噪声频率点,对待语音频率点和噪声频率点分别调整维纳滤波器的参数在滤除噪声的同时对语音频率点尽可能的保留从而减少语音失真。实验结果表明,与现有的双麦克风消噪系统相比,本系统在对噪声进行抑制的同时能够有效减少消噪算法对语音造成的损害,提高了手机的通话质量,对于方向性的语音干扰也能起到很好的抑制效果。     

Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.     

A cross-spectrum weighting algorithm for speech enhancement and array processing: combining phase-shift information and stationary signal properties

In this paper, a gain function for noise cancellation with a two-channel microphone array is presented. This gain function combines ideas from one- and multichannel algorithms. It is developed using a minimum mean square error estimator for the amplitude of the speech signal from the cross spectrum between two microphone signals. To consider speech pauses and the absence of spectral components of the speech, an extension of this gain function is presented. The performance of the overall gain function is shown in terms of the cancellation of (diffuse) driving noise as well as the cancellation of an interfering speech signal, both recorded in a car.     

A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones

This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.     

Model-based speech enhancement using a bone-conducted signal

Codebook-based single-microphone noise suppressors, which exploit prior knowledge about speech and noise statistics, provide better performance in nonstationary noise. However, as the enhancement involves a joint optimization over speech and noise codebooks, this results in high computational complexity. A codebook-based method is proposed that uses a reference signal observed by a bone-conduction microphone, and a mapping between air- and bone-conduction codebook entries generated during an offline training phase. A smaller subset of air-conducted speech codebook entries that accurately models the clean speech signal is selected using this reference signal. Experiments support the expected improvement in performance at low computational complexity.     

一阶指向可调差分阵列的前后向比分析与优化设计

一阶指向可调差分传声器阵列具有尺寸小、主瓣指向灵活可调、以及阵列响应不随频率变化等优点,因而在音频处理领域得到了重要关注。但差分传声器阵列对阵元失配误差较为敏感,在实际设计中需要予以考虑。前后向比直接反映了传声器阵列对后向噪声干扰的抑制程度,是差分传声器阵列设计中的重要指标。本文从理论上深入剖析了失配误差对一阶指向可调差分传声器阵列前后向比性能的影响,揭示了其中蕴含的规律。在此基础上,针对实际中同时存在随机失配误差情况,提出了一种最差性能优化设计方法。仿真实验和实测实验验证了本文理论分析的正确性和所提优化设计方法的有效性。      

基于时间反转的复杂声场拾声传声器阵列性能研究

探讨时间反转技术在复杂声场传声器阵列拾声中应用的可行性及其机理,给出其一般规律和性能。研究表明:在自由空间中,其拾声性能与频率,阵列形状和半径有关,频率越高,半径越大,拾声效果越好。在普通房间中,在语音频段内,圆弧阵列在预定目标点处的阵列增益性能要比离预定目标点约25 cm远处的位置处大5 dB以上。在普通房间和混响室中的实验验证了上述结论。      

A hybrid microphone array post-filter in a diffuse noise field

In this paper, a hybrid post-filter for microphone arrays with the assumption of a diffuse noise field is proposed to suppress correlated as well as uncorrelated noise. In the proposed post-filter, a modified Zelinski post-filter, which is estimated using the signals on the microphone pairs on which noises are uncorrelated by considering the correlation characteristics of noise impinging on different microphone pairs, is applied to the high frequencies to suppress spatially uncorrelated noise; a single-channel Wiener post-filter is applied to the low frequencies for cancellation of spatially correlated noise. In theory, the proposed post-filter is a Wiener post-filter. In practice, experiments using multi-channel recordings were conducted, and experimental results demonstrate the usefulness and superiority of the proposed post-filter compared to other post-filters using speech quality measures and speech recognition rate.     

采用球谐分解的二范数广义逆波束形成

基于传声器阵列的声成像技术是解决噪声源识别的有效途径之一.该文提出了一种基于球谐分解的L2范数广义逆波束形成算法,并对此算法在分布式球形阵列布放方案下进行了定位精度及鲁棒性的对比分析研究.仿真结果显示,此算法对低频相干声源具有较高的空间定位精确度,且阵元位置误差对此算法性能的影响有限.通过在半消声室进行实验进一步证明了...     

Cochannel speech separation

The multisignal minimum-cross-entropy spectral analysis (multisignal MCESA) is applied to the problem of separating the speech signals of two talkers speaking simultaneously on a single channel, e.g., when two talkers use a single microphone. A new two-stage approach to the problem is proposed in which a spectral separator is followed by a spectral tailoring procedure. The spectral separator produces an initial estimate of the speech spectrum for each talker. Then the spectral tailoring procedure employs the multisignal MCESA technique to adjust the initial spectral estimates to account for the characteristics of the known cochannel composite speech signal. The research emphasis is placed on the implementation and evaluation of the spectral tailoring procedure, i.e., the use of the multisignal MCESA in the proposed scheme. Its usefulness is evaluated and validated by listening tests and by comparing the spectral distortions of the estimated voices before and after the multisignal MCESA processing.     

球谐域自适应混响抵消与声源定位算法

提出了一种基于球谐域的自适应混响抵消与声源定位算法,该方法通过去混响处理改善语音质量,并提高球谐域定位算法在混响环境下的定位性能。推导了基于多通道线性预测的自适应混响抵消算法在球谐域的表达式,针对刚球模型提出分阶处理的去混响方法,并对去混响后的信号进行波达方向估计。采用32元球阵的仿真结果表明,相比于球谐域不分阶去混响方法,该方法最大可减少约2/3的运算量,同时语音PESQ得分及SRMR均显著提高。利用实验数据对算法性能进行测试,实验结果验证了该方法在实际声学环境中去混响和声源定位的有效性。      

Analysis with binaural auditory model and experiment on the timbre of Ambisonics recording and reproduction

A scheme for analyzing the timbre in spatial sound with binaural auditory model is proposed and the Ambisonics is taken as an example for analysis.Ambisonics is a spatial sound system based on physical sound field reconstruction.The errors and timbre colorations in the final reconstructed sound field depend on the spatial aliasing errors on both the recording and reproducing stages of Ambisonics.The binaural loudness level spectra in Ambisonics reconstruction is calculated by using Moore's revised loudness model and then compared with the result of real sound source,so as to evaluate the timbre coloration in Ambisonics quantitatively.The results indicate that,in the case of ideal independent signals,the high-frequency limit and radius of region without perceived timbre coloration increase with the order of Ambisonics.On the other hand,in the case of recording by microphone array,once the high-frequency limit of microphone array exceeds that of sound field reconstruction,array recording influences little on the binaural loudness level spectra and thus timbre in final reconstruction up to the highfrequency limit of reproduction.Based on the binaural auditory model analysis,a scheme for optimizing design of Ambisonics recording and reproduction is also suggested.The subjective experiment yields consistent results with those of binaural model,thus verifies the effectiveness of the model analysis.     

Ambisonics声捡拾与重放音色的双耳听觉模型分析与实验

提出用双耳听觉模型对空间声音色进行分析的普遍方法,并以Ambisonics为例进行了分析。Ambisonics是基于物理声场重构的空间声系统,其最终重构声场误差以及音色改变是由传声器捡拾和重放空间混叠误差共同引起的。采用修正的Moore双耳响度模型计算了Ambisonics重构声场的双耳响度级谱并和目标声场的情况比较,从而定量评价重构声场的音色改变。结果表明,在理想捡拾信号的情况下,无音色改变重放的上限频率和区域大小随Ambisonics的阶数而增加。而对于传声器阵列捡拾的情况,只要阵列的上限频率大于Ambisonics重放的上限频率,在重放的上限频率以下,传声器阵列空间混叠误差对最终重构声场及其感知音色的影响就可以忽略。在此基础上,提出了一种综合考虑捡拾与重放性能的Ambisonics系统优化设计方法。心理声学实验得到了和双耳听觉模型一致的结果,从而也验证了模型分析的有效性。      

Design of robust broadband beamformers using worst-case performance optimization for time-domain spherical microphone arrays

Spherical array modal beamformers are known to be sensitive to microphone mismatches. To combat the problem, robust design of spherical broadband beamformers with the time-domain implementation structure using the Worst-Case Performance Optimization(WCPO) is studied. It is shown that the conservativeness of the upper error bound on beamforming response is the primary factor to degrade the performance of the existing WCPO-based approach. Then an improved design approach for spherical broadband beamformers is presented using a stricter upper error bound on beamforming response. Theoretical analysis shows that the cost function of the proposed design is smaller and the range of feasible set of the proposed design is also larger when compared with the existing design. Moreover, a scheme on constraint specification has been proposed to reduce fluctuation of beamforming response over frequency at steering direction to alleviate signal distortion. Simulation results show that the proposed design outperforms its existing counterparts and exhibits a lower sidelobe level under the same condition of microphone mismatches.