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。     

Turbo迭代下的改进谱减法研究

自迭代式的谱减法虽然能有效提高去噪度,但是随着迭代次数的增加会造成语音畸变。结合Turbo迭代原理,在噪声子空间引入小波包多阈值去噪对迭代谱减法进行改进。将谱减法和小波包多阈值去噪根据Turbo原理结合起来,两个分量滤波算法各自工作在信号子空间和噪声子空间提取语音信号,并在每次迭代时将各自输出作为另一分量滤波算法的先验信息。仿真实验表明：该算法在10次Turbo迭代处理后收敛,并在低信噪比下能较好地去噪和降低语音失真度。     

基于小波包分解的含噪语音时频特性分析 及端点检测

针对Hilbert—Huang变换方法在语音处理过程中存在模态混叠问题,本文提出了基于小波包分解的语音时频分析方法。首先对含噪语音进行小波包分解,对各分量分别进行经验模态分解,并运用相关系数阈值准则对固有模态函数进行筛选;然后建立语音信号的Hilbert谱和瞬时能量谱;最后将基于小波包分解的Hilbert—Huang变换瞬时能量谱方法应用于含噪语音的端点检测。实验结果表明：与传统广义维数以及谱熵算法相比,本文方法具有更好的准确性、稳定性和自适应性,能够有效描述语音信号非线性非平稳的时频特性,为语音信号的分析研究提供了一个新的思路。     

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.     

强背景噪声下语音端点检测的算法研究

多带谱熵法对语音频段进行分带处理形成新的分带谱熵函数,在低信噪比时,该方法能够更好地检测出语音,还能体现能量分布情况,应用较为广泛。多窗谱分析方法对同一数据序列用多个正交的数据窗分别求直接谱,是一种低方差、高分辨率的谱分析方法,尤其适合非线性系统中高噪声背景下弱信号、时频演变信号的分析。提出基于多窗谱及多带谱相结合的语音检测方法,仿真结果表明：改进算法较其他算法占有绝对的优势,而且性能稳定。     

基于感知滤波器与统计方法的语音增强新算法

根据人耳感知特性,提出了一种小波包感知滤波器与统计方法相结合的语音增强新算法.小波包感知滤波器根据人耳Bark域频率感知特性,将含噪语音频带划分成24个频率群,每个频率群内信号进行最小均方误差对数谱幅度(MMSE-LSA)的估计.通过估计各频率群的先验信噪比得到待估计语音与含噪语音的增益方程,从而得到该频率群内的估计语音,最后将所得的分段估计语音重建即得到增强后的语音.实验结果表明,在各种噪声情况下,该方法均优于其他方法.     

基于多窗谱相关加权语音增强

与传统的周期谱图相比,多窗谱分析方法是一种低方差、高分辨率的谱分析方法,尤其适合非线性系统中强噪声背景下弱信号、时频演变信号的分析.基于多窗谱估计这种优点,提出多窗谱相关加权语音增强方法,先对噪声与含噪信号比(NNSR)进行估计,用基于 NNSR 的幅度谱减实现预语音增强,再用相关加权规则获得最终的增强语音.通过客观和...     

基于正交小波包分解的语音去噪增强

对带噪语音信号进行增强,是语音信号处理中一个重要的研究课题.由于噪声影响语音质量,这抑制背景噪声,利用小波包良好的时频分析能力,能较好模拟人耳基底膜频率分析特性的特点,提出基于正交小波包的语音去噪增强算法,算法首先把含噪语音信号分解于不同的频率范围内,根据"3σ规则",确定不同频率下的阈值,并采用动态阈值法对各层进行阈值处理,最后对处理后的语音信号反变换得到增强后的信号.在MATLAB平台上,对带噪语音信号去噪增强,实验结果表明,方法在抑制白噪声的同时减小了语音信息的损失,输出语音在保证可懂度的同时达到了较好的输出语音效果.     

基于小波包分解和噪声分析的抗噪说话人识别特征参数

为提高说话人识别中语音特征参数对噪声的鲁棒性,本文提出在对语音进行小波包分解基础上,分析噪声的特性,在不同子带内进行谱减并设立权重,提出了一种新的语音特征参数多层美尔倒谱系数.仿真实验表明,与MFCC特征参数相比,ML-MFCC在噪声环境下具有更好的抗噪性能和说话人识别率.     

Bark子带小波包自适应阈值语音去噪方法

为了克服低信噪比输入下,语音增强造成清音弱分量损失,导致信号重构失真的问题,提出了一种新的语音增强方法。该方法采用小波包拟合语音感知模型的临界带,按子带能量对语音清浊音分离,然后对清音和浊音信号分别作8层和4层小波包分解,在阈值计算上采用Bark子带小波包自适应节点阈值算法,在Bark子带实时跟踪噪声水平,有效保护清音中高频弱分量,减少失真。通过与传统语音增强方法的仿真对比实验,证实该方法在低信噪比输入时,具有明显优势,输出信噪比高,语音失真度低。将该方法与谱减法相结合,进行语音二次增强,能进一步提高增强语音质量。     

A new optimum feature extraction and classification method for speaker recognition: GWPNN

Speech and speaker recognition is an important topic to be performed by a computer system. In this paper, an expert speaker recognition system based on optimum wavelet packet entropy is proposed for speaker recognition by using real speech/voice signal. This study contains both the combination of the new feature extraction and classification approach by using optimum wavelet packet entropy parameter values. These optimum wavelet packet entropy values are obtained from measured real English language speech/voice signal waveforms using speech experimental set. A genetic-wavelet packet-neural network (GWPNN) model is developed in this study. GWPNN includes three layers which are genetic algorithm, wavelet packet and multi-layer perception. The genetic algorithm layer of GWPNN is used for selecting the feature extraction method and obtaining the optimum wavelet entropy parameter values. In this study, one of the four different feature extraction methods is selected by using genetic algorithm. Alternative feature extraction methods are wavelet packet decomposition, wavelet packet decomposition – short-time Fourier transform, wavelet packet decomposition – Born–Jordan time–frequency representation, wavelet packet decomposition – Choi–Williams time–frequency representation. The wavelet packet layer is used for optimum feature extraction in the time–frequency domain and is composed of wavelet packet decomposition and wavelet packet entropies. The multi-layer perceptron of GWPNN, which is a feed-forward neural network, is used for evaluating the fitness function of the genetic algorithm and for classification speakers. The performance of the developed system has been evaluated by using noisy English speech/voice signals. The test results showed that this system was effective in detecting real speech signals. The correct classification rate was about 85% for speaker classification.     

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.     

基于改进能熵比的维纳滤波语音增强算法

为了提高低信噪比环境下语音增强的效果、算法的鲁棒性.在基于维纳滤波算法的基础上,结合基于频域特征的语音端点检查算法,提出了一种新的语音增强算法.端点检测算法使用小波包ERB子带的谱熵和改进的频域能量的能熵比法.其中,小波包ERB子带的谱熵考虑了人耳听觉掩蔽模型和语音与噪声信号之间的频率分布之间的不同;频域能量利用了有语音帧和无语音帧的能量不同.维纳滤波算法实时采集语音数据并使用新的参数来区别无语音段和有语音段,并在无语音段平滑更新噪声谱.实验结果表明,该端点检测算法能够很好的区分有语音段和无语音段,这就使得在低信噪比的情况下语音增强效果得到了提升,同时算法的鲁棒性和实时性也得到了保障.在与其他两种算法对比中,得到了更好的语音增强效果.     

基于小波包的语音谱熵端点检测方法研究

端点检测是语音识别过程中的一个重要的环节,因此改善端点检测的效果一直是语音识别领域的一个重要课题。为了提高在背景噪声下语音信号端点检测的准确率,提出了一种基于小波包的谱熵端点检测方法。该方法对语音信号进行小波包变换,将每帧信号分解成多个子带,在此基础上计算每帧信号的子带能量,通过子带能量所占比例求出每帧信号的谱熵,最后确定新的门限值。仿真实验表明,该方法比传统方法更有效、更优越,能够比较准确地检测语音信号。     

基于SPIHT编码的语音信号压缩算法

提出了一种基于最佳小波包变换和SPIHT编码的语音信号压缩编码方法。该方法首先对语音信号进行小波包变换,求解最佳小波树,进行动态位分配,再用改进的SPIHT算法对变换后的小波系数进行压缩编码。并且采用了熵编码的方法进一步提高了压缩比。实验表明,该方法在较高的压缩比下能获得较好的信号重构质量,计算复杂度低,延迟小。     

基于改进谱减方法的语音增强研究

由于噪声的影响导致语音信号的质量降低,因此需要对语音信号进行语音增强。语音增强是语音信号处理的前沿领域,其主要目标足从带噪语音中提取纯净的原始语音信号。介绍了实现语音增强方法的原理,利用实验仿真了传统谱减法和改进谱减方法,改进法通过对带噪信号进行参数调整,然后进行频域谱减,实验结果表明改进方法对语音增强效果明显好于传统方法。此外,对传统谱减法和改进谱减法的信噪比分别进行了计算,结果表明改进谱减方法的信噪比相对传统谱减方法有很大提高。