基于自适应仿生小波变换的语音增强方法

分析自适应滤波和小波滤波的原理与方法,提出一种基于自适应仿生小波变换的语音增强方法.该方法首先用仿生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,然后将经仿生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.通过选用自适应滤波器的最小二乘算法(RLS)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.实验结果表明,该方法对语音信号有显著的增强效果,能实现语音信号在同频段对噪声成分和有用信号的最佳估计.     

一种时间自适应阈值的小波包改进语音增强算法

针对传统小波语音增强算法存在过度周值处理的问题,提出一种改进的时间自适应阈值小波包去噪算法.该方法采用听觉感知小波包对噪声语音进行分解,得到小波包听觉感知节点上的系数,并基于语音存在概率估计按帧自动调节去噪周值,因改进的闲值能更好地避免语音小波包系数被过度阈值处理的情况,从而在抑制噪声的同时保留了更多的原始语音成分,进一步提高了降噪效果,实验结果表明,该算法比常规小波自适应闻值算法能得到更清晰的语音增强信号.     

一种时间自适应阈值的小波包改进语音增强算法

针对传统小波语音增强算法存在过度阈值处理的问题,提出一种改进的时间自适应阈值小波包去噪算法.该方法采用听觉感知小波包对噪声语音进行分解,得到小波包听觉感知节点上的系数,并基于语音存在概率估计按帧自动调节去噪阈值.因改进的阈值能更好地避免语音小波包系数被过度阈值处理的情况,从而在抑制噪声的同时保留了更多的原始语音成分,进一步提高了降噪效果.实验结果表明,该算法比常规小波自适应阈值算法能得到更清晰的语音增强信号.

     

基于小波包与自适应维纳滤波的语音增强算法

语音增强主要用来提高受噪声污染的语音可懂度和语音质量,它的主要应用与在嘈杂环境中提高移动通信质量有关。传统的语音增强方法有谱减法、维纳滤波、小波系数法等。针对复杂噪声环境下传统语音增强算法增强后的语音质量不佳且存在音乐噪声的问题,提出了一种结合小波包变换和自适应维纳滤波的语音增强算法。分析小波包多分辨率在信号频谱划分中的作用,通过小波包对含噪信号作多尺度分解,对不同尺度的小波包系数进行自适应维纳滤波,使用滤波后的小波包系数重构进而获取增强的语音信号。仿真实验结果表明,与传统增强算法相比,该算法在低信噪比的非平稳噪声环境下不仅可以更有效地提高含噪语音的信噪比,而且能较好地保存语音的谱特征,提高了含噪语音的质量。     

基于小波与小波包的变速器故障特征提取

本文结合自适应小波变换滤波去噪方法与小渡阈值去噪方法,提出了一种可用于变速器故障振动信号去噪的双层滤波去噪算法.该算法的滤波过程分为两层,第一层滤波采用自适应小波变换滤波算法;第二层滤波采用经典的小波阈值去噪算法对信号进行二次去噪.最后,将去噪后的故障信号采用小波包进行了分解,并提取了小波包频带能量作为故障特征向量.     

自适应语音信息隐藏的小波方法

语音信号作为一种特殊的信息隐藏载体,在其中进行信息隐藏时必须充分考虑人类的听觉特性。该文以小波变换为基础,提出了一种盲检测的自适应语音信息隐藏算法。该算法首先依据语音信号小波变换的系数来刻画不同语音段对人类听觉感知特性的影响,然后根据这种特征刻画来确定秘密信息的嵌入位置和强度,同时兼顾了鲁棒性和隐藏量的要求,最后以数字实验验证了该算法具有较好的信息隐藏性能。     

基于仿生小波变换和自适应阈值的语音增强方法

提出一种基于仿生小波变换以及自适应阈值的语音增强方法．含噪语音通过仿生小波变换后,针对不同的尺度采用不同的阈值函数进行去噪．由于在小波变换过程中考虑了人耳的听觉特性,所提出方法优于基本的小波语音增强方法．实验表明,该方法在多种噪声条件下均具有较好的语音增强效果．     

自适应语音分段增强算法

近年来小波理论在信号分析、处理中得到了广泛的应用,本文提出了一种自适应分段小波域语音增强(ASWE)算法,即采用局部余弦包变换对语音信号自适应分段,然后对每一段语音采用基于小波变换的语音增强处理该方法不需要噪声的先验知识,且适合于缓慢变化的非平稳噪声.最后的仿真实验表明,该方法比直接用小波去噪效果好,是一种有效的语音增强技术     

一种基于小波变换图像去噪的方法

提出了一种基于图像软阈值小波变换的高斯白噪声消除法。该算法根据含噪声图的特点,把信号分成信号象素与可能噪声象素两类,对于可能是噪声的象素,采用图像的小波软阈值去噪方法进行滤波,而对信号象素不产生影响,且能保留更多的图像细节。文中也给出了标准中值滤波,自适应维纳滤波算法和小波软阈值去噪的算法进行比较实验,结果表明用小波软阈值去噪的算法处理高度污染高斯白噪声的图像能力明显强于标准中值滤波,稍微优于自适应维纳滤波算法,且能够比较好保留图像的细节部分。     

基于子带谱熵的仿生小波语音增强

前端噪声处理直接关系着语音识别的准确性和稳定性,针对小波去噪算法所分离出的信号不是原始信号的最佳估计,提出一种基于子带谱熵的仿生小波变换(BWT)去噪算法。充分利用子带谱熵端点检测的精确性,区分含噪语音部分和噪声部分,实时更新仿生小波变换中的阈值,精确地区分出噪声信号小波系数,达到语音增强目的。实验结果表明,提出的基于子带谱熵的仿生小波语音增强方法与维纳滤波方法相比,信噪比(SNR)平均提高约8%,所提方法对噪声环境下语音信号有显著的增强效果。     

基于小波熵自适应阈值的语音信号去噪新方法

针对语音信号去噪问题, 提出小波熵自适应阈值去噪法。首先利用小波变换分解带噪语音信号, 计算小波分解后信号子带区间的小波熵, 然后将小波熵和自适应阈值相结合确定各层高频系数的阈值门限, 采用折中指数阈值函数对各层高频系数进行去噪处理, 重构降噪后的语音信号, 最后对比小波熵自适应阈值、极大极小阈值、固定阈值和无偏风险阈值去噪方法的性能。实验结果表明, 当输入信噪比为5 dB时, 小波熵自适应阈值去噪法的输出信噪比是最大的, 且其输入输出信噪比曲线高于其他三种阈值去噪法的输入输出信噪比曲线, 从而证实该算法具有更好的去噪性能。     

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．     

基于改进的小波阈值函数语音增强方法

传统的小波阈值去噪方法会造成有用语音信号的损失, 信噪比改善情况不理想. 通过分析小波去噪原理, 提出了一种改进的小波阈值函数语音增强方法. 该方法结合小波软、硬阈值函数去噪的优点, 克服了硬阈值函数的不连续及软阈值函数存在偏差的缺点. 该方法首先对清浊音信号进行判断, 接着采用变化的阈值对清浊音信号的小波系数进行不同的阈值处理. 仿真实验结果表明, 改进的方法非常适用于强噪声背景下的语音增强, 无论在保留含噪语音信号中的清音信息, 还是在信噪比改善指标上均优于传统的软阈值法、谱减法和听觉感知小波变换法.     

基于小波调制尺度的语音特征参数提取方法

时频分析的理论基础上,提出了一种基于小波调制尺度特征的参数提取方法。根据人对调制谱信息的感知特性及干扰在调制谱中的特点,采用小波分析技术及归一化处理求得归一化的小波调制尺度特征参数,并以此作为语音的动态特征应用于语音识别系统。通过与MFCC一阶、二阶系数对比的汉语音节识别实验表明,该方法在抗噪声干扰和说话速率变化等方面比MFCC的一阶、二阶系数的性能优越,为提高语音识别鲁棒性提供了一种新途径。     

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.     

Speech enhancement using Teager energy operated ERB-like perceptual wavelet packet decomposition

In recent past, wavelet packet (WP) based speech enhancement techniques have been gaining popularity due to their inherent nature of noise minimization. WP based techniques appeared as more robust and efficient than short-time Fourier transform based methods. In the present work, a speech enhancement method using Teager energy operated equal rectangular bandwidth (ERB)-like WP decomposition has been proposed. Twenty four sub-band perceptual wavelet packet decomposition (PWPD) structure is implemented according to the auditory ERB scale. ERB scale based decomposition structure is used because the central frequency of the ERB scale distribution is similar to the frequency response of the human cochlea. Teager energy operator is applied to estimate the threshold value for the PWPD coefficients. Lastly, Wiener filtering is applied to remove the low frequency noise before final reconstruction stage. The proposed method has been applied to evaluate the Hindi sentences database, corrupted with six noise conditions. The proposed method’s performance is analysed with respect to several speech quality parameters and output signal to noise ratio levels. Performance indicates that the proposed technique outperforms some traditional speech enhancement algorithms at all SNR levels.     

Speech enhancement based on undecimated wavelet packet-perceptual filterbanks and MMSE–STSA estimation in various noise environments

In this paper, we proposed a new speech enhancement system, which integrates a perceptual filterbank and minimum mean square error–short time spectral amplitude (MMSE–STSA) estimation, modified according to speech presence uncertainty. The perceptual filterbank was designed by adjusting undecimated wavelet packet decomposition (UWPD) tree, according to critical bands of psycho-acoustic model of human auditory system. The MMSE–STSA estimation (modified according to speech presence uncertainty) was used for estimation of speech in undecimated wavelet packet domain. The perceptual filterbank provides a good auditory representation (sufficient frequency resolution), good perceptual quality of speech and low computational load. The MMSE–STSA estimator is based on a priori SNR estimation. A priori SNR estimation, which is a key parameter in MMSE–STSA estimator, was performed by using “decision directed method.” The “decision directed method” provides a trade off between noise reduction and signal distortion when correctly tuned. The experiments were conducted for various noise types. The results of proposed method were compared with those of other popular methods, Wiener estimation and MMSE–log spectral amplitude (MMSE–LSA) estimation in frequency domain. To test the performance of the proposed speech enhancement system, three objective quality measurement tests (SNR, segSNR and Itakura–Saito distance (ISd)) were conducted for various noise types and SNRs. Experimental results and objective quality measurement test results proved the performance of proposed speech enhancement system. The proposed speech enhancement system provided sufficient noise reduction and good intelligibility and perceptual quality, without causing considerable signal distortion and musical background noise.