ICA和线性神经网络在带噪语音分离中的研究

用基于独立分量分析（ICA）的盲源分离方法对强噪声背景下的混合语音信号进行分离时,如果忽略噪声的影响则会产生很差的分离效果。为克服此不足,结合噪声对消和盲源分离,提出了一种在强噪声背景环境下的混合语音分离方法,即先将带噪观测信号通过线性神经网络构成自适应噪声对消器,然后采用ICA进行分离,与增加一路噪声作为源信号的分离方法相比,该方法具有更好的分离效果。     

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,其主要目标是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.     

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.     

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。     

基于curvelet变换和独立分量分析的含噪盲源分离

独立分量分析（ICA）是基于信号高阶统计量的盲源分离方法,在高阶统计量方法中,由于高斯信号的高阶累计量为零,所以系统存在加性高斯噪声时就难以处理。提出了一种基于curvelet阈值去噪和FastICA算法的含噪信号盲分离的方法,并对高斯噪声环境下的混合图像进行了盲分离的仿真。结果表明,该方法能很好地解决由于存在加性高斯噪声而导致经典ICA算法性能发生严重恶化的问题;同时将curvelet变换去噪应用于含噪图像的盲源分离中,可以提高混合图像的信噪比,相对于小波去噪后的ICA算法,其分离性能有很大改善。     

基于听觉掩蔽效应的多频带谱减语音增强方法

为了减小传统谱减法引入的音乐噪声,提出了一种将多频带谱减和听觉掩蔽效应相结合的语音增强算法.用加权递归平滑的方法估计噪声的功率谱,对带噪的语音信号进行多频带谱减,计算听觉掩蔽阈值,再根据掩蔽阈值动态地调节谱减因子,通过增益函数得到增强后语音信号的频谱.仿真实验结果表明,与传统的谱减法相比,该算法在信噪比较低情况下,背景噪声和残余噪声得到了有效的抑制,语音信号的清晰度和可懂度也有了明显提升.     

基于独立分量分析的单通道语音增强算法

传统的独立分量分析要求观测信号的个数不能小于源信号的个数,无法直接对单路信号进行独立分量分析。为了能够利用独立分量分析分离加性噪声,须构造一路观测信号。基于语音信号的短时平稳的特性,该文提出一种构造噪声信号的算法,实现了信号与噪声的分离。仿真结果表明,利用该算法可得到很好的消噪结果,提高信号的信噪比。     

独立分量分析在有噪图像分离中的应用

独立分量分析(independent component analysis，ICA)是基于信号高阶统计量的盲源分离方法。在分析独立分量分析的基本模型及方法的基础上，讨论了有噪信号的独立分量分析(Noisy ICA)，利用小波阈值去噪和FastICA算法进行了有噪混合图像分离的仿真研究。结果表明，对于含有加性观测噪声的混合图像的分离，先去噪处理再进行独立分量分离的效果要优于独立分量分离后再去噪的效果。     

基于改进谱平滑策略的IMCRA算法及其语音增强

噪声谱估计算法在单通道语音增强方法中起着重要作用,为了改善噪声谱估计算法对噪声的估计和更新能力,结合最小统计（MS）算法,对改进的基于控制的递归平均（IMCRA）噪声谱估计算法的递归平均参数进行改进,并用一阶递归的方式对平滑功率谱的最小值进行改进。采用谱减法对含噪语音信号作去噪处理,从客观和主观两方面对不同算法的性能进行评价,对比分析不同噪声不同信噪比下增强前后语音的分段信噪比（segSNR）、PESQ得分、MOS得分。实验结果表明,提出的方法能够更好地跟踪噪声信号变化,改善语音质量。     

基于谱减法的改进语音增强方法

语音增强技术是解决噪声污染的一项强有力的预处理技术.谱减法通过处理后的语音中会留下所谓的"音乐噪声",针对这个问题,提出了一种多带谱相减与感觉加权相结合的语音增强方法.对带噪语音进行多带谱相减,并根据人的听觉掩蔽特性,对多带谱相减后的信号进行感觉加权,从而进一步降低背景噪声.在语音失真和噪声抑制之间取得良好的折中,减少语音的听觉失真,有效地抑制"音乐噪声",提高语音的清晰度.     

基于自适应参数估计的鲁棒语音增强算法

基于鲁棒H_∞滤波器理论和共轭梯度自适应参数估计方法提出了一种对复杂噪声有抑制效果的语音增强算法。应用这种方法自适应地从带噪信号中提取语音参数时不必预先知道噪声源的统计特性,只要求噪声信号能量有限。因为它基于H_∞滤波器,所以可保证由外界干扰和附加噪声引起的性能指标恶化达到最小。仿真结果表明：该语音增强算法具有计算速度快、鲁棒性好、语音增强效果明显、易于实现、可抑制复杂背景噪声等特点。     

Phase-based dual-microphone robust speech enhancement

A dual-microphone speech-signal enhancement algorithm, utilizing phase-error based filters that depend only on the phase of the signals, is proposed. This algorithm involves obtaining time-varying, or alternatively, time-frequency (TF), phase-error filters based on prior knowledge regarding the time difference of arrival (TDOA) of the speech source of interest and the phases of the signals recorded by the microphones. It is shown that by masking the TF representation of the speech signals, the noise components are distorted beyond recognition while the speech source of interest maintains its perceptual quality. This is supported by digit recognition experiments which show a substantial recognition accuracy rate improvement over prior multimicrophone speech enhancement algorithms. For example, for a case with two speakers with a 0.1 s reverberation time, the phase-error based technique results in a 28.9% recognition rate gain over the single channel noisy signal, a gain of 22.0% over superdirective beamforming, and a gain of 8.5% over postfiltering.     

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.     

Noisy speech processing by recurrently adaptive fuzzy filters

Two noisy speech processing problems-speech enhancement and noisy speech recognition-are dealt with. The technique we focus on is by using the filtering approach; a novel filter, the recurrently adaptive fuzzy filter (RAFF), is proposed and applied to these two problems. The speech enhancement is based on adaptive noise cancellation with two microphones, where the RAFF is used to eliminate the noise corrupting the desired speech signal in the primary channel. As to the noisy speech recognition, the RAFF is used to filter the noise in the feature domain of speech signals. The RAFF is inherently a recurrent multilayered connectionist network for realizing the basic elements and functions of dynamic fuzzy inference, and may be considered to be constructed from a series of dynamic fuzzy rules. As compared to other existing nonlinear filters, three major advantages of the RAFF are observed: 1) a priori knowledge can be incorporated into the RAFF, which makes the fusion of numerical data and linguistic information possible; 2) owing to the dynamic property of the RAFF, the exact lagged order of the input variables need not be known in advance; 3) no predetermination, like the number of hidden nodes, must be given since the RAFF can find its optimal structure and parameters automatically Several examples on adaptive noise cancellation and noisy speech recognition problems using the RAFF are illustrated to demonstrate the performance of the RAFF     

An Information-Theoretic View of Array Processing

The removal of noise and interference from an array of received signals is a most fundamental problem in signal processing research. To date, many well-known solutions based on second-order statistics (SOS) have been proposed. This paper views the signal enhancement problem as one of maximizing the mutual information between the source signal and array output. It is shown that if the signal and noise are Gaussian, the maximum mutual information estimation (MMIE) solution is not unique but consists of an infinite set of solutions which encompass the SOS-based optimal filters. The application of the MMIE principle to Laplacian signals is then examined by considering the important problem of estimating a speech signal from a set of noisy observations. It is revealed that while speech (well modeled by a Laplacian distribution) possesses higher order statistics (HOS), the well-known SOS-based optimal filters maximize the Laplacian mutual information as well; that is, the Laplacian mutual information differs from the Gaussian mutual information by a single term whose dependence on the beamforming weights is negligible. Simulation results verify these findings.     

Speech enhancement using MMSE estimation under phase uncertainty

Most of the speech enhancement algorithms process the amplitudes of speech, but the phase of noisy speech is left unprocessed as it may cause undesired artifacts. Recently, short time Fourier transform based single channel speech enhancement algorithms are developed by considering uncertain prior knowledge of phase. The uncertain knowledge of the phase is obtained from the phase reconstruction algorithms. The goal of this paper is to develop joint minimum mean square error estimate of complex speech coefficients given uncertainty phase (CUP) information by considering Nagakami probability density function (PDF) and gamma PDF as speech spectral amplitude priors and generalized gamma PDF for noise prior. Estimators like amplitudes given uncertainty phase, which uses uncertain phase only for amplitude estimation and not for phase improvement are developed. Experimental results shows that incorporating uncertain phase information improves quality and intelligibility of speech. Also novel phase-blind estimators are developed using Nagakami PDF/gamma as speech priors and generalized gamma as noise prior. Finally comparison of all estimators using uncertain prior phase information is discussed and how initial phase information affects the enhancement process is analyzed with novel estimators. For comparison of all the derived estimators, the speech signals uttered by male and female speakers are taken from TIMIT database. The proposed CUP estimators outperforms the existing algorithms in terms of objective performance measure segmental signal to noise ratio, phase signal to noise ratio, perceptual evaluation of speech quality, short time objective intelligibility.     

盲源分离在单通道语音增强算法中的应用

提出一种单通道语音增强算法。首先由接收到的单声道语音信号的含噪部分构造一个假想噪声源,将这一噪声源和含噪的信号作为多通道自适应去相关（MAD）盲分离算法的输入,得到增强的语音信号。进一步将这一增强的语音作为输入,利用Daubechies小波对其进行分解,在小波域中选取合适的阈值函数进行滤波,然后合成时域语音信号。根据以上步骤得到的增强语音有较高的信噪比及可懂度。     

Harmonic retrieval by period blind source extraction method: Model and algorithm

A frequently encountered problem in signal processing field is harmonic retrieval in additive colored Gaussian or non-Gaussian noise, especially when the frequencies of the harmonic signals are closely spaced in frequency domain. The purpose of this paper is to develop novel harmonic retrieval algorithm based on blind source extraction (BSE) method from linear mixtures of harmonic signals using only one observed channel signal. First, we establish the blind source separation (BSS) based harmonic retrieval model in additive noise using the only one observed channel, at the same time, the fundamental principle of BSE based harmonics retrieval algorithm is analyzed in detail. Then, based on the established harmonic BSS model, we propose a BSE approach to the harmonic retrieval using the concept of period BSE method, as a result, the harmonic retrieval algorithm using only one channel mixed signals is derived. Simulation results show that the proposed algorithm is able to separate the harmonic source signals and yield ideal performance.     

滤波器组自适应语音增强方法

双麦克风噪声抵消应用中,由于交叉串的存在,传统自适应算法降噪性能受到很大的影响。为了提高双麦克风算法降噪性能,使用两级自适应滤波系统消除交叉串扰问题。为提高自适应滤波器收敛性能,采用主从结构LMS算法自适应调节步长因子。同时为了适合窄带处理算法,将输入信号进行子带分析预处理,对每个子带独立进行抗交叉串绕自适应处理,将各子带增强信号合并得到增强语音信号。实验结果表明,该方消噪量大,语音损伤小,语音增强效果显著。