一种自适应变阶谱减降噪算法在电子耳蜗中的应用

由于传统谱减语音增强存在残留的"音乐噪声",因此基于传统谱减法降噪的电子耳蜗(CI)感知的声音品质也会受到影响.为提高CI的抗噪性,本文提出了一种自适应变阶谱减算法,并将该方法应用于电子耳蜗的语音增强中.根据CI电极对应的频带关系,该算法先对采集的带噪声音信号功率谱进行Bark子带划分,并在每个Bark子带中根据信噪比的变化进行谱减阶数和系数的自适应调节,使各子带噪声更均衡地去除,基本消除了传统方法存在的"音乐噪声".基于该算法的电子耳蜗ACE仿真实验及测听结果表明,与传统谱减法相比,改进的算法能更好地抑制背景噪声和残留噪声,仿真得到的CI合成音感知更好和更清晰.     

基于多尺度总体最小二乘的图像去噪

提出了一种基于多尺度总体最小二乘的图像去噪算法.采用平稳小波变换对噪音图像进行分解,分别对各个分解层的高频子带,通过总体最小二乘算法估计信号小波系数;并且考虑到不同尺度小波系数之间的相关性,将尺度相关性约束到总体最小二乘算法中,进而准确估计各高频子带信号小波系数,再由估计的信号小波系数通过小波逆变换得到去噪图像.实验结果表明,考虑尺度间相关性的总体最小二乘平稳小波变换图像去噪算法能有效去除图像噪音,在信噪比和视觉质量上有了较大改善.     

基于多尺度总体最小二乘的图像去噪

提出了一种基于多尺度总体最小二乘的图像去噪算法.采用平稳小波变换对噪音图像进行分解,分别对各个分解层的高频子带,通过总体最小二乘算法估计信号小波系数|并且考虑到不同尺度小波系数之间的相关性,将尺度相关性约束到总体最小二乘算法中,进而准确估计各高频子带信号小波系数,再由估计的信号小波系数通过小波逆变换得到去噪图像.实验结果表明,考虑尺度间相关性的总体最小二乘平稳小波变换图像去噪算法能有效去除图像噪音,在信噪比和视觉质量上有了较大改善.     

基于多窗谱的心理声学语音增强

与传统的周期谱图相比,多窗谱具有更小的估计方差。从含噪语音的多窗谱对噪声及噪声与含噪语音之比(NNSR)进行估计,用基于NNSR的幅度谱减实现用于计算人耳掩蔽阈值的预增强语音,用集成了人耳掩蔽阈值的心理声学加权规则实现最终的增强语音。考虑到多窗谱的特点对掩蔽偏移量进行了修正,修正后的重建语音,其客观测量指标修正巴克谱测度比修正前有一定的改进。再对心理声学加权规则作最大值小于1的限制,则输入信噪比越大(0 dB以上),分段信噪比和总体信噪比提高得越多。非正式试听表明重建语音失真较小,背景噪声大大降低,且没有音乐噪声。     

稳健的子带子阵级导向最小方差波束形成算法

导向最小方差(STMV)波束形成是一种利用导向协方差矩阵获得自适应权值的方法,具有快速收敛特性.常规的稳健导向最小方差(RSTMV)波束形成算法在处理宽频带信号时,性能下降明显.为了改善算法的性能,结合频域子带划分和空域子阵划分技术,提出一种多子带不确定集独立约束的稳健子阵级STMV波束形成算法。通过频域子带划分可对不同子带的导向向量误差范数边界进行约束,计算出各子带对应的对角加载量,得到稳健的子带级最小方差波束形成算法权向量;同时采用子阵技术进行降维处理,可进一步增加划分子带的数目,从而提高算法的性能并有效降低计算复杂度,最终得到一种稳健的子带子阵级STMV波束形成算法。理论分析和仿真结果表明,在阵列导向向量存在误差的情况下,该算法在干扰方向形成的零陷最深,且零陷波束宽度最窄,输出信噪比接近理论值,因此性能最佳.实际海试数据处理表明,在强干扰目标存在时,弱目标输出信干噪比较RSTMV算法可提高4 dB,较常规波束形成可提高10 dB,在角度分辨力和算法复杂度方面得到有效改善,同时可以保证目标功率无失真输出。      

小波包自适应阈值语音降噪新算法

为了克服低信噪比输入下,语音增强造成语音清音中的弱分量损失,造成重构信号包络失真的问题。论文提出了一种新的语音增强方法。该方法根据语音感知模型,采用不完全小波包分解拟合语音临界频带,并对语音按子带能量进行清浊音区分处理,在阈值计算上,提出了一种清浊音分离,基于子带信号能量的小波包自适应阈值算法。通过仿真实验,客观评测和听音测试表明,该算法在低信噪比输入时较传统算法,能够更加有效地减少重构信号包络失真,在不损伤语音清晰度和自然度的前提下,使输出信噪比明显提高。将该算法与能量谱减法结合,进行二次增强能进一步提高降噪输出的语音质量。     

子带t分布的快速独立向量分析在语声盲源 分离中的应用*

为了提高独立向量分析算法在盲语声分离任务中的分离性能,降低算法计算复杂度,并改善目前尚未完全解决的顺序模糊性的问题,该文提出一种基于子带t分布的快速独立向量分析算法。在声源模型方面,该算法首先利用语声信号重尾分布的特性,假设声源概率密度函数服从t分布,同时采用子带建模的方法来增强同一声源相邻频点的高阶依赖性,进而减轻频点间的顺序不一致问题。在空间模型方面,该算法采用秩1更新的方式估计声源信号,避免矩阵求逆操作和分离矩阵的估计,从而降低计算复杂度。实验结果表明,与现有的基于独立向量分析的盲源分离算法相比,该算法能够在相同的迭代次数下取得更优的语声分离性能。     

用于语音增强的高频信噪比度量

一般使用信噪比的提高评估带噪语音的增强算法效果,本文提出一种新的信噪比计算方法——高频信噪比（SNRH）,即用经过高通滤波的各信号计算信噪比。实验表明新方法比传统方法能更准确地评估语音增强算法的好坏和有效范围,是更合理的语音增强算法度量。     

基于FFT奇异值分解的光谱信号去噪算法

微型光谱仪在采集光谱信号过程中,光谱数据经常受到来自仪器光学系统和电子电路中的干扰出现噪声和光源特征峰,严重干扰了真实光谱信号的图谱特征,因此需要使用合理的预处理方法保留光谱信号中有用信号并尽可能过滤噪声信号同时将光源特征峰滤除,从而提高光谱信息定量分析的稳健性和准确性。并且在线检测系统要求尽可能减少人为参数选择对去噪效果的影响,奇异值分解经常应用于由系统电路引起的噪声去噪,奇异值降噪阶次的选取对提高信号信噪比十分关键,但是往往参数选取主要依赖经验调试和实验验证。因此,提出了一种基于奇异值重构信号分量频率的光谱信号去噪算法。该算法首先重构原始光谱信号单个奇异值分量信号,然后对每个奇异值分量信号作快速傅里叶变换,得到每个奇异值分量信号快速傅里叶变换结果中振幅最大所对应的频率值,最后按照奇异值递减方式对相应分量信号频率值进行一阶滞后差分,得到频率差分谱,研究表明,差分谱第一个谱峰值在大于设定阈值处所对应的奇异值即为奇异值分解降噪的有效阶次。结果表明：对包含多种重金属离子的溶液在线测量的紫外可见光谱信号,添加不同强度的随机噪声,并进行去噪处理,使用信噪比和均方根误差两个性能指标进行对比。所提算法相较于SG滤波算法和小波变换去噪算法信噪比分别提高了22.05%,10.88%,均方根误差分别降低了74.28%,41.29%。所提算法完全基于数据驱动,在处理真实紫外可见光谱信号中不仅抑制了噪声影响,而且将微型光谱仪的光源特征峰有效滤除,在紫外可见光谱信号的定量分析中具有较好的应用前景。     

基于二代小波变换的红外图像非线性增强算法

红外图像具有对比度低和信噪比低等特点,实用中必须进行增强处理.将小波分析与模糊逻辑相结合,提出了一种基于二代小波变换的红外图像非线性增强算法.该算法首先利用二代小波变换对图像进行分解,提取图像的多尺度细节特征,然后,根据目标和背景噪声信号的差异,通过模糊非线性增强算子分别对各个分解层的高频子带进行非线性增强来改变目标特征的强度,抑制背景信号,最后利用小波反变换重构图像,以实现图像的对比度增强和背景抑制.与几种常用的图像增强算法实验结果相比,此算法能有效地抑制图像中的背景噪声,增强目标内容信息,取得了较好的增强效果.     

Noise reduction of speech signals using time-varying and multi-band adaptive gain control for smart digital hearing protectors

In this paper, a single-channel speech enhancement algorithm based on non-linear and multi-band Adaptive Gain Control (AGC) is proposed. The algorithm requires neither Signal-to-Noise Ratio (SNR) nor noise parameters estimation. It reduces the background noise in the temporal domain rather than the spectral domain using a non-linear and automatically adjustable gain function for multi-band AGC. The gain function varies in time and is deduced from the temporal envelope of each frequency band to highly compress the frequency regions where noise is present and lightly compress the frequency regions where speech is present. Objective evaluation using the PESQ (Perceptual Evaluation of Speech Quality) metric shows that the proposed algorithm performs better than three benchmarks, namely: the spectral subtraction, the Wiener filter based on a priori SNR estimation and a band-pass modulation filtering algorithm. In addition, blind subjective tests show that the proposed algorithm introduces less musical noise compared to the benchmark algorithms and was preferred 78.8% of the time in terms of signal quality. The proposed algorithm is implemented in a miniature low power digital signal processor to validate its feasibility and complexity for smart hearing protection in noisy environments.     

Dual-channel spectral subtraction algorithms based speech enhancement dedicated to a bilateral cochlear implant

In this paper, two speech enhancement algorithms (SEAs) based on spectral subtraction (SS) principle have been evaluated for bilateral cochlear implant (BCI) users. Specifically, dual-channel noise power spectral estimation algorithm using power spectral densities (PSD) and cross power spectral density (CPSD) of the observed signals was studied. The enhanced speech signals were obtained using either Dual Channel Non Linear Spectral Subtraction ‘DC-NLSS’ or Dual-Channel Multi-Band Spectral Subtraction ‘DC-MBSS’ algorithms. For performance evaluation, some objective speech assessment tests relying on Perceptual Evaluation of Speech Quality (PESQ) score and speech Itakura-Saito (IS) distortion measurement were performed to fix the optimal number of frequency band needed in DC-MBSS algorithm. In order to evaluate the speech intelligibility, subjective listening tests were assessed with 50 normal hearing listeners using a specific BCI simulator and with three deafened BCI patients. Experimental results, obtained using French Lafon database corrupted by an additive babble noise at different Signal-to-Noise Ratios (SNR), showed that DC-MBSS algorithm improves speech understanding better than DC-NLSS algorithm for single and multiple interfering noise sources.     

Speech enhancement by array crosstalk resistant ANC and spectrum subtraction

Microphone array-based speech enhancement has great importance for speech communications and speech recognition. To reduce the aperture of the microphone array and to increase the effect of the speech enhancement will greatly broaden the application areas of the microphone array. An array crosstalk resistant adaptive noise cancellation method is therefore presented. And then an improved spectral subtraction algorithm is further cascaded to obtain better enhancement results. Theoretic analysis and experiments indicate that the proposed scheme needs only a very small microphone array while it simultaneously achieves a higher SNR improvement. Besides, the proposed scheme can be used in many noisy environments and is easy for real-time implementation.     

Noise-robust voice conversion based on joint dictionary optimization

A noise robust voice conversion algorithm based on joint dictionary optimization is proposed to effectively convert noisy source speech into the target one. In composition of the joint dictionary, speech dictionary is optimized using backward elimination algorithm. At the same time, a noise dictionary is introduced to match the noisy speech. The experimental results show that the backward elimination algorithm can reduce the number of dictionary frames and reduce the amount of calculation while ensuring the conversion effect. In low SNR and multiple noise environments, the algorithm has better conversion effect than both the traditional NMF algorithm and the NMF conversion algorithm plus spectral subtraction de-noising. The proposed algorithm improves the robustness of voice conversion system.     

Envelope expansion methods for speech enhancement

The last decade has seen increasing interest in techniques for the enhancement of digital speech signals. Significant gains have been made in terms of signal-to-noise ratio (SNR) and quality, but few techniques have produced improvements in intelligibility. A method for speech enhancement based on nonlinear expansion of the spectral envelope is presented. The expansion is consistent with both the long-term spectrum of the speech and with the probability that speech is present in a given sample. Objective SNR measures are used to compare this algorithm with the well-known spectral subtraction method, with an alternative expansion scheme, and with limiting SNRs resulting from perfect recovery of the amplitude spectrum. For the purpose of intelligibility assessments, a simplified version of the algorithm has been implemented on a Texas Instruments TMS320-C25 system. Listening trials with this real-time system, conducted using a modified rhyme test, have produced small, but consistent, improvements in articulation scores.     

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.     

Perceptually motivated wavelet packet transform for bioacoustic signal enhancement

A significant and often unavoidable problem in bioacoustic signal processing is the presence of background noise due to an adverse recording environment. This paper proposes a new bioacoustic signal enhancement technique which can be used on a wide range of species. The technique is based on a perceptually scaled wavelet packet decomposition using a species-specific Greenwood scale function. Spectral estimation techniques, similar to those used for human speech enhancement, are used for estimation of clean signal wavelet coefficients under an additive noise model. The new approach is compared to several other techniques, including basic bandpass filtering as well as classical speech enhancement methods such as spectral subtraction, Wiener filtering, and Ephraim-Malah filtering. Vocalizations recorded from several species are used for evaluation, including the ortolan bunting (Emberiza hortulana), rhesus monkey (Macaca mulatta), and humpback whale (Megaptera novaeanglia), with both additive white Gaussian noise and environment recording noise added across a range of signal-to-noise ratios (SNRs). Results, measured by both SNR and segmental SNR of the enhanced wave forms, indicate that the proposed method outperforms other approaches for a wide range of noise conditions.     

采用联合字典优化的噪声鲁棒性语音转换算法

针对含噪语音难以实现有效的语音转换,本文提出了一种采用联合字典优化的噪声鲁棒性语音转换算法。在联合字典的构成中,语音字典采用后向剔除算法(Backward Elimination algorithm,BE)进行优化,同时引入噪声字典,使得含噪语音与联合字典相匹配。实验结果表明,在保证转换效果的前提下,后向剔除算法能够减少字典帧数,降低计算量。在低信噪比和多种噪声环境下,本文算法与传统NMF算法和基于谱减法消噪的NMF转换算法相比具有更好的转换效果,噪声字典的引入提升了语音转换系统的噪声鲁棒性。      

An approach based on simplified KLT and wavelet transform for enhancing speech degraded by non-stationary wideband noise

It is well known that the non-stationary wideband noise is the most difficult to be removed in speech enhancement. In this paper a novel speech enhancement algorithm based on the dyadic wavelet transform and the simplified Karhunen-Loeve transform (KLT) is proposed to suppress the non-stationary wideband noise. The noisy speech is decomposed into components by the wavelet space and KLT-based vector space, and the components are processed and reconstructed, respectively, by distinguishing between voiced speech and unvoiced speech. There are no requirements of noise whitening and SNR pre-calculating. In order to evaluate the performance of this algorithm in more detail, a three-dimensional spectral distortion measure is introduced. Experiments and comparison between different speech enhancement systems by means of the distortion measure show that the proposed method has no drawbacks existing in the previous methods and performs better shaping and suppressing of the non-stationary wideband noise for speech enhancement.