一种基于改进谱减法的语音增强方法

谱减法是最常用的一种语音增强技术,其特点是计算复杂度低、实时性强、易于实现.谱减法的主要目的是去除语音信号中的噪声干扰,提高语音信号质量.本文在研究基于改进谱减法的基础之上,提出了利用带噪语音的高频区估计噪声谱以及由短时过零率和短时能量组合而成的加权函数去除背景噪声及音乐噪声的语音增强方法.实验表明,这种时频结合的语音增强方法对背景噪声下的语音质量的增强效果明显.     

一种基于宽带噪声的同态滤波语音增强算法分析

语音通信中,强烈的背景噪声会影响语音信号的传输质量,为了提高语音通信的抗噪声性能,针对说话时伴随着呼吸引起的宽带噪声,分析传统抗噪声方法存在的问题,提出自相关与同态滤波相结合的改进方法,对纯净的语音信号和噪声信号进行分离,提取纯净语音信号.通过实验仿真明显提高了语音处理系统的传输质量,达到了语音增强的目的,其鲁棒性更好.     

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.     

一种基于仿生小波变换的语音增强方法

提出了一种新的基于仿生小波变换的语音增强方法。该方法通过对仿生小波变换系数进行阈值处理,从而达到语音增强的目的。实验结果表明：该方法在四种实际噪声环境下均优于一些经典方法如：谱减法、维纳滤波和基于离散小波变换的阈值去噪方法,具有更好的语音增强效果。     

一种基于盲源提取的子空间语音增强算法

本文提出了一种单麦克风下的间接语音增强算法.该算法基于两个重要模块：第一个模块,采用基于均方协差预测的盲源提取算法将附加噪声从嘈杂的语音信号中提取出来.第二个模块,利用了一种基于语音和附加噪声协方差矩阵的广义子空间方法,提取纯净的语音信号.对该算法进行了白噪声环境和嘈杂火车等真实环境噪声下的仿真实验.实验结果表明,提出的算法有良好的语音增强效果,性能上与其他算法比较有明显的优势.将算法应用于噪声环境下的语音识别处理中,很大程度地降低了噪声对语音识别的影响,取得了良好的识别率.     

语音与噪声的特性及其研究现状

在噪声环境下的语音识别率将会受到严重的影响.语音增强是解决噪声污染的有效方法.在语音增强技术中,语音识别和说话人识别是很重要的.因此.识别装置通常工作在环境噪声下.语音增强不仅与信号处理技术相关,并涉及到人的听觉感知和语音认知.由于噪声的来源有很多,在不同的应用场合,其特点也各不相同.因此很难确定一个通用的适用于各种环境噪声的语音增强算法.根据不同的噪声,采用不同的语音增强策略.     

随机共振在强噪声环境中语音增强应用

传统的语音增强方法是在保持语音可懂度和清晰度的前提下,尽可能地从带噪语音中提取需要的纯净语音,而在强噪声环境中,语音信号表现为弱信号,去噪变得困难.基于Hodgkin-Huxley神经元阈上非周期随机共振原理,提出一种自适应调节,添加最佳噪声来进行语音随机共振,从而实现语音增强.Matlab实验结果表明,在强噪声环境中实现对语音信号增强,信噪比提高明显,且效果优于传统算法.方法具有一定鲁棒性,提供了在强噪声环境中增强语音信号的新思路.     

基于高斯过程模型的语音增强

针对信号处理领域的语音活动探测问题,提出一种基于高斯过程先验假设的概率方法,用于增强语音。利用高斯过程模型的后验概率来估计纯净语音,使用在学习过程中得到的高斯过程模型的参数探测语音活动。实验结果表明,该方法对于在白噪声和有色噪声环境下的语音有较好的增强效果。     

基于三种估计准则下的语音增强方法

语音信号在产生、传输和接收过程中不可避免要受到各种噪声的干扰.为了提高语音清晰度和可懂度,减轻听觉疲劳,增强语音识别效率,需要对带噪声语音进行降噪处理.语音增强技术在语音通信和语音识别过程中有重要价值.在简要介绍语音增强技术的基础上,详细论述了联合最大后验概率估计准则、最大后验概率估计准则和最小均方误差估计准则下的频域语音增强方法的原理及特点,并提出了一种噪声谱估计方法,然后对几种语音增强方法进行了实验仿真.实验证明,基于最小均方误差估计准则的增强方法的效果最好,基于最大后验概率估计准则的增强效果较差.     

基于自适应滤波的语音增强和噪声消除

语音增强的主要目标是从带噪语音信号中提取尽可能纯净的原始语音.文中介绍了一种基于自适应滤波进行语音增强的方法,这种方法比其他方法多用了1个参考噪声作为辅助输入,从而获得了比较全面的关于噪声的信息,因而能得到更好的降噪效果.通过计算机上的模拟处理,处理后的语音信号较原噪声语音信号显著地提高了信噪比,同时能有效地改善语音可懂度.     

Multi notch adaptive digital filter design for enhancement of speech signals embedded in non-stationary noise

This paper presents a new approach to speech enhancement based on modified least mean square-multi notch adaptive digital filter (MNADF). This approach differs from traditional speech enhancement methods since no a priori knowledge of the noise source statistics is required. Specifically, the proposed method is applied to the case where speech quality and intelligibility deteriorates in the presence of background noise. Speech coders and automatic speech recognition systems are designed to act on clean speech signals. Therefore, corrupted speech signals by the noise must be enhanced before their processing. The proposed method uses a primary input containing the corrupted speech signal and a reference input containing noise only. The new computationally efficient algorithm is developed here based on tracking significant frequencies of the noise and implementing MNADF at those frequencies. To track frequencies of the noise time-frequency analysis method such as short time frequency transform is used. Different types of noises from Noisex-92 database are used to degrade real speech signals. Objective measures, the study of the speech spectrograms and global signal-to-noise ratio (SNR), segmental SNR (segSNR) as well as subjective listing test demonstrate consistently superior enhancement performance of the proposed method over tradition speech enhancement method such as spectral subtraction.     

稀疏低秩模型下的单通道自学习语音增强算法

针对现有基于字典学习的增强算法依赖先验信息的问题,基于矩阵的稀疏低秩分解提出一种无监督的单通道语音增强算法。该算法首先通过稀疏低秩分解将带噪语音的幅度谱分解为低秩、稀疏和噪声三部分,然后通过对低秩部分进行自学习构建出噪声字典,最后利用所得噪声字典和乘性迭代准则于低秩和稀疏部分中分离出纯净语音。相较于其他基于字典学习的语音增强算法,本文所提算法无需语音或噪声的先验信息,因而更加方便和实用。实验结果显示,本文算法能够在保留语音谐波结构的同时有效抑制噪声,增强效果明显优于鲁棒主成分分析和多带谱减法。     

MCRA noise estimation for KLT-VRE-based speech enhancement

A new signal subspace-based approach is proposed for the enhancement of speech corrupted by a high level of noise. Conventional subspace-based methods use the minimum mean square error criterion to optimize the Karhunen-Loève Transform (KLT). In non-stationary noisy environments, the selection of the optimal order of the KLT-based speech enhancement model is a critical issue. Indeed, estimation of the relevant subspace dimensions depends on the environmental conditions that may change unpredictably. Therefore, a drastic KLT-based dimension reduction may induce the loss of relevant components of speech and conversely, a reconstruction using a higher order of the KLT model will be ineffective to remove the noise. The method presented in this paper uses a Variance of Reconstruction Error (VRE) criterion to optimally select the KLT order model. A prominent point of this subspace method is that it incorporates the Minima Controlled Recursive Averaging (MCRA) to estimate the noise Power Spectral Density (PSD) used in the gain function. Three variants of the VRE combined with MCRA methods are implemented and compared, namely the VRE-MCRA, VRE-MCRA2 and VRE-IMCRA. Objective measures show that VRE-based approaches achieve a lower signal distortion and a higher noise reduction than existing enhancement methods.     

FPGA multi-filter system for speech enhancement via multi-criteria optimization

Speech is the main medium for human communication and interaction. Apart from the traditional telephones, more and more applications come with speech interfaces, which use speech signal as an input for various purposes. However, many of these applications might fail to perform in noisy environments as the signal-to-noise ratio (SNR) degrades. Two important measures for any speech enhancement algorithm are noise suppression and speech distortion. Naturally, different speech enhancement algorithms will have different trade-offs. Moreover, depending on the environment, it is possible that one algorithm will outperform the others in some respects. This paper proposes a multi-filter system, which has the capability of continually adjusting the noise suppression level and the speech distortion level in a Pareto fashion. Moreover, we show that the system works under a variety of noisy environments and we obtain the efficient frontier of the combined filters for each background noise. Because the multi-filters are adapting in parallel, the final system can be implemented on FPGA efficiently.     

An iterative longest matching segment approach to speech enhancement with additive noise and channel distortion

This paper presents a new approach to speech enhancement from single-channel measurements involving both noise and channel distortion (i.e., convolutional noise), and demonstrates its applications for robust speech recognition and for improving noisy speech quality. The approach is based on finding longest matching segments (LMS) from a corpus of clean, wideband speech. The approach adds three novel developments to our previous LMS research. First, we address the problem of channel distortion as well as additive noise. Second, we present an improved method for modeling noise for speech estimation. Third, we present an iterative algorithm which updates the noise and channel estimates of the corpus data model. In experiments using speech recognition as a test with the Aurora 4 database, the use of our enhancement approach as a preprocessor for feature extraction significantly improved the performance of a baseline recognition system. In another comparison against conventional enhancement algorithms, both the PESQ and the segmental SNR ratings of the LMS algorithm were superior to the other methods for noisy speech enhancement.     

Noisy Speech Enhancement Using Harmonic-Noise Model and Codebook-Based Post-Processing

This paper presents a post-processing speech restoration module for enhancing the performance of conventional speech enhancement methods. The restoration module aims to retrieve parts of speech spectrum that may be lost to noise or suppressed when using conventional speech enhancement methods. The proposed restoration method utilizes a harmonic plus noise model (HNM) of speech to retrieve damaged speech structure. A modified HNM of speech is proposed where, instead of the conventional binary labeling of the signal in each subband as voiced or unvoiced, the concept of harmonicity is introduced which is more adaptable to the codebook mapping method used in the later stage of enhancement. To restore the lost or suppressed information, an HNM codebook mapping technique is proposed. The HNM codebook is trained on speaker-independent speech data. To reduce the sensitivity of the HNM codebook to speaker variability, a spectral energy normalization process is introduced. The proposed post-processing method is tested as an add-on module with several popular noise reduction methods. Evaluations of the performance gain obtained from the proposed post-processing are presented and compared to standard speech enhancement systems which show substantial improvement gains in perceptual quality     

基于语音增强失真补偿的抗噪声语音识别技术

本文提出了一种基于语音增强失真补偿的抗噪声语音识别算法。在前端,语音增强有效地抑制背景噪声;语音增强带来的频谱失真和剩余噪声是对语音识别不利的因素,其影响将通过识别阶段的并行模型合并或特征提取阶段的倒谱均值归一化得到补偿。实验结果表明,此算法能够在非常宽的信噪比范围内显著的提高语音识别系统在噪声环境下的识别精度,在低信噪比情况下的效果尤其明显,如对-5dB的白噪声,相对于基线识别器,该算法可使误识率下降67.4%。     

Optimal speech enhancement under signal presence uncertainty using Log Gabor Wavelet and Bayesian Joint Statistics

This paper investigates the problem of speech enhancement when only a single microphone is used and the statistics of the interfering noise and speech are not available a priori. Thus it seeks to address a pitfall of many current enhancement techniques and look towards a system which would have application in the real world. This paper focuses on Log Gabor Wavelet (LGW) based Long Term Squared Spectral Amplitude estimator using the Maximum a Posteriori (MAP) criterion. To begin with, long term cepstral mean subtraction technique with LGW is proposed to suppress telephone channel and handset effect from the speech signals. Then a novel speech enhancer by MAP based Bayesian Bivariate Model is developed to suppress the background noise. This work also introduces an inter-scale dependency between the coefficients and their parents by a Circularly Symmetric probability density function related to the family of Spherically Invariant Random Process (SIRPs). The corresponding joint estimator is derived by MAP estimation theory. The inter-scale noise variance of the coefficients is kept constant which gives closed form solution. Consideration of speech presence uncertainty (SPU) estimator is another contribution to the proposed estimator. Therefore, in this paper, the main contributions are; (i) combination of LGW, SIRPs and SPU for background noise reduction, (ii) LGW and Long Term Cepstral Mean Subtraction to reduce the effects of both telephone channel and handsets, (iii) circularly Symmetric probability density function to exploit the inter-scale dependency between the coefficients and their parents and corresponding joint estimators are derived by MAP estimation theory, (iv) the inter-scale noise variance of the coefficients is kept constant which gives closed form solution, (v) idea refines the estimate of the magnitudes by scaling them by the SPU probability. Extensive comparisons are done among the proposed and existing speech enhancement algorithms on NOIZEUS speech database which has different types of noise. We report the subjective and objective evaluations encompassing four classes of algorithms: spectral subtractive, subspace, statistical model based and Wiener type against the proposed methods. Experimental results show that the proposed estimator yields a higher improvement in Segmental SNR (SSNR), lower Log Area Ratio (LAR), Weighted Spectral Slope (WSS) distortion, higher Perceptual Evaluation of Speech Quality (PESQ) and Mean Opinion Score (MOS) compared to the existing speech enhancement algorithms. For SSNR measure, the proposed methods show 2 dB of improvement than existing methods for almost every Noise sources. For MOS measure, the proposed methods show improvement than existing methods for almost every Noise sources. Therefore the proposed methods are aiming to enhance the speech quality as well as intelligibility at a time.     

改进相位谱补偿的语音增强方法

针对传统单通道语音增强方法中用带噪语音相位代替纯净语音相位重建时域信号，使得语音主观感知质量改善受限的情况，提出了一种改进相位谱补偿的语音增强算法。该算法提出了基于每帧语音输入信噪比的Sigmoid型相位谱补偿函数，能够根据噪声的变化来灵活地对带噪语音的相位谱进行补偿；结合改进DD的先验信噪比估计与语音存在概率算法（SPP）来估计噪声功率谱；在维纳滤波中结合新的语音存在概率噪声功率谱估计与相位谱补偿来提高语音的增强效果。相比传统相位谱补偿（PSC）算法而言，改进算法可以有效抑制音频信号中的各类噪声，同时增强语音信号感知质量，提升语音的可懂度。