The optimized wavelet filters for speech compression

In this paper, optimized wavelet filters for speech compression are proposed whose wavelet filter coefficients are derived with different window techniques such as Kaiser and Blackman windows via simple linear optimization. When the developed wavelet filters are exploited for speech compression, they not only give better compression ratio but also yield good fidelity parameters as compared to other wavelet filters. A comparative study of performance of different existing wavelet filters and the proposed wavelet filters is made in terms of compression ratio (CR), signal-to-noise ratio (SNR), peak signal-to-noise ratio (PSNR) and normalized root-mean square error (NRMSE) at different thresholding levels. The simulation result included in this paper shows increased efficacy and improved performance of the proposed filters in the field of speech signal processing.     

基于小波变换的语音同步方法研究

为提高语音通信干扰效果客观评估中标准语音文件和受扰语音文件的同步精度,对军标GJB4405A-2006中规定的标准语音文件增设了高低频交叉的双音频同步头;介绍了小波消噪的原理,利用2层sym小波对受扰语音文件进行了消噪处理,将同步头中的高频信号作为噪声进行大幅削弱;设计了基于幅度比较的同步算法,找出同步基准点,并对20组受扰严重的语音文件的进行了仿真测试,仿真结果证明了该方法得到的同步精度符合后续数据处理的要求。     

小波阈值降噪算法中最优分解层数的自适应选择

小波阚值降噪算法是一种去除数字信号中白噪声的有效算法．针对加性高斯白噪声的情况,提出一种自适应小波降噪算法,用于语音信号的增强．它能根据带噪信号的特点,自适应选择小波变换的最优分解层数．实验结果表明,该算法比经典的小波降噪算法具有更好的降噪效果,能有效提高算法的实用性能．     

基于分频带最优阈值函数的电力信号去噪

研究电力信号在传输过程中受到噪声污染问题。由于电网负荷较大,系统的运行随机性强,系统非线性引起电力信号噪声严重。针对小波阈值去噪算法存在较大的缺陷,传统改进算法全局阈值去噪的局限性,提出了一种适合电力信号去噪的分频带(分层)最优阈值函数去噪方法。新方法根据电力信号与噪声各频带能量分布特性得到各频带最优阈值,并对含噪电力信号逐层分频带最优阈值函数去噪处理,明显地降低了波形畸变率。仿真与实验结果表明,分频带最优阈值函数去噪方法明显提高了电力信号的去噪效果,提高了电力信号的检测精确度。     

矿用钢丝绳损伤检测信号处理方法研究

采用电磁检测法检测矿用钢丝绳受损情况时,检测信号中含有大量噪声,且存在尖峰和突变干扰,增大了损伤识别难度,需要对原始检测信号进行降噪处理。常用的傅里叶变换无法处理运行中的钢丝绳检测信号,而小波变换因存在平移不变性较差、频带混叠等问题而影响检测准确度。提出了基于双树复小波变换的矿用钢丝绳损伤检测信号处理方法。首先采用Q平移法构造双树复小波高低通滤波器,对原始信号进行3层双树复小波分解,得到高低频信号分量;然后采用最小极大方差软阈值方法对分解信号进行降噪处理;最后对降噪信号进行重构。在实验室环境下搭建了钢丝绳损伤检测试验平台,对基于双树复小波变换的钢丝绳损伤检测信号处理方法的降噪性能进行验证,结果表明:该方法可有效减少检测信号中的尖峰和突变数量,使信号平稳,降噪效果优于经典小波变换,且增大了奇异点处信号峰值,有利于后续特征提取。     

小波语音增强算法在说话人识别中的应用

干净语音环境下识别率很高的说话人识别系统,在有噪声语音环境下识别性能显著降低。针对这一问题,将小波语音增强算法应用于说话人识别系统,提出一种结点阈值去噪新方法。语音增强主要目的是从带噪语音中尽可能地提取纯净的原始语音。在不同信噪比条件下进行实验,结果表明,提出的方法比传统的阈值法能更好地提高语音质量。     

A wavelet- based transform method for quality improvement in noisy speech patterns of Arabic language

This paper addresses the problem of single-channel speech enhancement of low (negative) SNR of Arabic noisy speech signals. For this aim, a binary mask thresholding function based coiflet5 mother wavelet transform is proposed for Arabic speech enhancement. The effectiveness of binary mask thresholding function based coiflet5 mother wavelet transform is compared with Wiener method, spectral subtraction, log-MMSE, test-PSC and p-mmse in presence of babble, pink, white, f-16 and Volvo car interior noise. The noisy input speech signals are processed at various levels of input SNR range from ?5 to ?25 dB. Performance of the proposed method is evaluated with the help of PESQ, SNR and cepstral distance measure. The results obtained by proposed binary mask thresholding function based coiflet5 wavelet transform method are very encouraging and shows that the proposed method is much helpful in Arabic speech enhancement than other existing methods.     

Audio enhancement using local SNR-based sparse binary mask estimation and spectral imputation

This paper proposes a method for enhancing speech and/or audio quality under noisy conditions. The proposed method first estimates the local signal-to-noise ratio (SNR) of the noisy input signal via sparse non-negative matrix factorization (SNMF). Next, a sparse binary mask (SBM) is proposed that separates the audio signal from the noise by measuring the sparsity of the pool of local SNRs from the adjacent frequency bands of the current and several previous frames. However, some spectral gaps remain across frequency bands after applying the binary masks, which distorts the separated audio signal due to spectral discontinuity. Thus, a spectral imputation technique is used to fill the empty spectrum of the frequency band where it is removed by the SBM. Spectral imputation is conducted by online learning NMF with the spectra of the neighboring non-overlapped frequency bands and their local sparsity. The effectiveness of the proposed enhancement method is demonstrated on two different tasks use speech and musical content, respectively. Consequently, objective measurements and subjective listening tests show that the proposed method outperforms conventional speech and audio enhancement methods, such as SNMF-based alternatives and deep recurrent neural networks for speech enhancement, block thresholding, and a commercially available software tool for audio enhancement.     

Image quality based comparative evaluation of wavelet filters in ultrasound speckle reduction

This paper presents the comparative study of various wavelet filter based denoising methods according to different thresholding values applied to ultrasound images. An original image is transformed into a multi scale wavelet domain and the wavelet coefficients are processed by a soft thresholding method. The denoised image is the output image obtained from the inverse wavelet transform of the threshold coefficients using Donoho's method. It has been observed that such denoising methods are effective in the sense that they preserve the edge details besides suppressing the noise. The comparative evaluation of the denoising performance is shown using statistical significance tests for different wavelet filters. Image quality parameters such as peak signal-to-noise ratio, normalized mean square error, and correlation coefficient have been used to evaluate the performance of wavelet filters. The performance has also been compared with the adaptive weighted median filtering method.     

小波变换的频响特性及其在语音去噪中的应用

讨论小波变换在实际语音信号去噪处理中应用。由于语音信号的复杂性 ,信号本身含有奇异性 ,因此不能单一使用阈值去噪法。文中定义了小波变换频响特性 ,并利用它重构低尺度参数上的小波变换模极大 ,达到去噪目的。实例证明它的有效性     

基于自适应阈值的脑电信号去噪方法

脑电采集后得到的脑电信号(Electroencephalogram,EEG)中含有噪声信号,为了有效去除噪声并保留有用信息,本文在软阈值去噪的基础上,提出一种改进阈值去除EEG噪声的算法。利用小波变换对EEG信号分解,得到多层的高频系数和低频系数;根据分解层次不同,对小波系数进行自适应的阈值处理;将缩放后的小波系数重构,得到去噪后的EEG信号。以信噪比、均方根误差作为去噪效果的定量指标,将改进算法与硬阈值法、软阈值法、Garrote阈值法进行比较,结果表明,改进阈值法优于其他3种阈值法。     

小波包分解下的多窗谱估计语音增强算法

传统谱减法是基于短时傅里叶变换的单一分辨率算法,具有较大方差。为此,提出一种基于小波包分解下的多窗谱估计语音增强算法。将含噪语音在小波包下分解成不同频段,在不同频段下进行多窗谱谱减运算,并逐一进行小波包重构,以得到去噪后的语音信号。仿真结果表明,该算法能提高含噪语音的信噪比,降低语言失真度。     

OM-LSA和小波阈值去噪结合的语音增强

针对OM-LSA(optimally modified log-spectral amplitude estimator)算法产生的残留噪声,提出了一种结合OM-LSA和小波阈值去噪的语音增强算法。首先,进行语音对数幅度谱估计;然后,估计残留噪声,利用带噪语音第一级小波系数和语音不存在时的增益函数进行估计,解决了常规方法对增强后语音噪声估计不准确的问题;最后,在小波域利用软阈值法对语音信号进行阈值处理。实验结果表明,提出的算法有效地去除了OM-LSA算法中的残余噪声,在分段信噪比(segmental signal-to-noise ratio,SegSNR)和对数谱失真(log-spectral distortion,LSD)等指标评价上有较大的提高。     

一种基于听觉特性的自适应小波语音增强方法

提出一种基于人类听觉特性的自适应小波滤波算法。该方法用听觉感知小波变换对含噪语音信号进行小波分解,这样可以保证对信号频率和幅值的听觉特性,将经听觉感知小波变换所分离出来的噪声成分作为自适应滤波器的输入。通过采用递推最小二乘算法从而实现信噪分离的最佳滤波,以保证去除信号中的相关噪声。结果表明,该方法能实现非平稳信号在同频段对噪声成分和有用信号的最佳估计,提高了语音的清晰度和可懂度。     

基于小波变换的高光谱图像消噪

本文主要针对高光谱图像的特点，利用波段间的几何信息高冗余性，通过小波分解去除高频的噪声和几何信息，保留低频的光谱信息。以其他波段的几何信息辅助噪声污染波段重构，经过相应的小波重构滤波器滤波，获得该波段图像的重建以进行消噪。     

A generalized time-frequency subtraction method for robust speech enhancement based on wavelet filter banks modeling of human auditory system.

We present a new speech enhancement scheme for a single-microphone system to meet the demand for quality noise reduction algorithms capable of operating at a very low signal-to-noise ratio. A psychoacoustic model is incorporated into the generalized perceptual wavelet denoising method to reduce the residual noise and improve the intelligibility of speech. The proposed method is a generalized time-frequency subtraction algorithm, which advantageously exploits the wavelet multirate signal representation to preserve the critical transient information. Simultaneous masking and temporal masking of the human auditory system are modeled by the perceptual wavelet packet transform via the frequency and temporal localization of speech components. The wavelet coefficients are used to calculate the Bark spreading energy and temporal spreading energy, from which a time-frequency masking threshold is deduced to adaptively adjust the subtraction parameters of the proposed method. An unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. Through rigorous objective and subjective evaluations, it is shown that the proposed speech enhancement system is capable of reducing noise with little speech degradation in adverse noise environments and the overall performance is superior to several competitive methods.     

A Generalized Time–Frequency Subtraction Method for Robust Speech Enhancement Based on Wavelet Filter Banks Modeling of Human Auditory System

We present a new speech enhancement scheme for a single-microphone system to meet the demand for quality noise reduction algorithms capable of operating at a very low signal-to-noise ratio. A psychoacoustic model is incorporated into the generalized perceptual wavelet denoising method to reduce the residual noise and improve the intelligibility of speech. The proposed method is a generalized time-frequency subtraction algorithm, which advantageously exploits the wavelet multirate signal representation to preserve the critical transient information. Simultaneous masking and temporal masking of the human auditory system are modeled by the perceptual wavelet packet transform via the frequency and temporal localization of speech components. The wavelet coefficients are used to calculate the Bark spreading energy and temporal spreading energy, from which a time-frequency masking threshold is deduced to adaptively adjust the subtraction parameters of the proposed method. An unvoiced speech enhancement algorithm is also integrated into the system to improve the intelligibility of speech. Through rigorous objective and subjective evaluations, it is shown that the proposed speech enhancement system is capable of reducing noise with little speech degradation in adverse noise environments and the overall performance is superior to several competitive methods.     

基于小波变换和数据融合技术的图像降噪方法

提出了一种基于小波变换和数据融合技术的图像降噪的方法．此方法对同一原始图像信号不同噪声的多源图像分别进行小波分解，在图像分解的高频域内，对小波系数进行阈值处理后，再进行数据融合处理，根据“多数原则”选择重要小波系数．在低频域内，新的逼近系数则通过对多幅图像的逼近系数直接进行加权平均得到．然后利用重要小波系数和逼近系数进行小波反变换，即可得到融合后的图像．实验结果表明：此方法既可以有效地降低噪声，又可以较好地保持图像细节．     

Enhancement of noisy speech based on a custom thresholding function with a statistically determined threshold

Performance of the thresholding based speech enhancement methods largely depend on the estimate of the exact threshold value as well as on the choice of the thresholding function. In this paper, a speech enhancement method is presented, in which a custom thresholding function is proposed and employed upon the Wavelet Packet (WP) coefficients of the noisy speech. The thresholding function is capable of switching between modified hard and semisoft thresholding functions depending on a parameter that decides the signal characteristics under consideration. Here, the threshold is determined based on the statistical modeling of the Teager energy operated WP coefficients of the noisy speech. Extensive simulations indicate that the threshold thus obtained in conjunction with the custom thresholding function is very effective in reduction of not only the white noise but also the color noise from the noisy speech thus resulting in an enhanced speech with better quality and intelligibility. Several standard objective measures and subjective evaluations including informal listening tests show that the proposed method outperforms the recent state-of-the-art thresholding based approaches of noisy speech enhancement from high to low levels of SNR.     

基于阈值的小波域语音增强新算法

提出了一种新的基于阈值的小波域语音增强算法,采用Bark尺度小波包对含噪语音进行分解,以模拟人耳的听觉特性.采用结点阈值法,用基于谱熵的方法估计结点噪声,实验表明,该算法在多种噪声,尤其是有色噪声和非平稳噪声条件下均有较好的语音增强效果.