一种新的用于语音识别的噪音估计及非线性谱相减方法

提出了一种新的噪音估计及非线性谱相减方法，通常的噪音估计一般基于语音检测方法，取噪音段的谱平均作为噪音谱的估计，该方法在信噪比较低时性能下降严重。提出的基于能量聚类的噪音谱估计方法，不依赖于语音检测直接估计噪音谱，提高了噪音谱估计的精度。还在一般非线性谱相减方法的基础上提出了改进的谱相减方法，该方法根据单个mel滤波器频带内局部的信噪比，来决定该频段内非线性谱相减的多少，细化了非线性差谱的额度，在有效抑制噪声的同时减少了语音谱的失真。     

基于TEO能量谱减法的语音增强技术的研究

谱减法是目前减少噪声干扰、提高语音质量的一种有效方法。为了进一步提高谱减法的去噪性能,提出一种基于TEO(Teager Energy Operator)能量的改进谱减法。该方法利用TEO能量对带噪语音进行语音活动检测,区别出噪声段和语音段,对噪声段和语音段分别进行谱减处理,既保证了语音质量,又尽可能地消除了噪声干扰。在F16战斗机噪声环境中对算法性能进行测试,结果表明,该方法提高了输出信噪比,抑制了音乐噪声,具有良好的语音增强效果。     

基于码本学习的改进谱减语音增强算法

提出一种可适应非平稳噪声环境的基于码本学习的改进谱减语音增强算法。该算法分为训练阶段和增强阶段。训练阶段,使用自回归模型对语音和噪声的频谱形状进行建模并构造语音和噪声码本;增强阶段,采用对数谱最小化算法估计出语音和噪声的频谱,通过谱相减消除噪声。算法在每个时间帧估计语音和噪声频谱,即使在语音存在时仍能够有效跟踪快速变化的非平稳噪声;采用自回归模型能得到噪声频谱的平滑估计,减少了音乐噪声。实验仿真表明,相比于传统谱减法和多带谱减法,改进的谱减法具有更好的噪声抑制性能并且语音失真更小。     

一种基于快速噪声估计的MMSE语音增强算法

提出一种基于快速噪声估计的MMSE语音增强算法,实验表明这种算法比起谱相减法和基于语音短时对数谱的最小均方误差（MMSE-LSA）算法能更显著地提高算法的客观性能,在非平稳噪声环境中能快速估计出变化的噪声功率谱。     

一种改进型谱减算法的语音增强研究

针对低信噪比条件下基本谱减算法存在降噪效果不佳,产生音乐噪声过大,语音可懂度不高的问题,提出了一种改进型的谱减算法。算法先计算语音信号的倒谱距离值,检测出噪音段和语音段,用动态计算的噪声值代替基本谱减法采用的噪声统计均值;根据当前帧和噪声帧的倒谱距离比值动态设置谱减系数,改进了传统算法中谱减系数保持不变的缺点;同时采用三种方法抑制音乐噪声。仿真实验表明,在低信噪比情况下,改进型的谱减算法可以有效降噪,提高信噪比和可懂度,达到语音增强的目的。     

多窗谱估计的语音增强减法研究

多带谱熵不仅能体现和谱熵一样的频率特性,还能体现能量的分布情况,因此在进语音检测时更趋向于采用多带谱熵估计。通过仿真,证明多带谱熵估计在非平稳信号检测中相比于谱熵估计的优越性,确定适合坦克环境的多带谱熵噪声估计算法。结合多带谱熵估计、相关加权、分帧相减等理论,提出一种以多窗谱估计为基础的改进的语音增强算法。仿真结果表明,提出的算法不仅能更好地抑制背景噪声和音乐噪声,而且还较好地保持了语音的可懂度和自然度。     

基于自适应噪声估计的语音增强技术

针对语音系统受外界强噪声干扰而导致识别精度降低以及通信质量受损的问题,提出一种基于自适应噪声估计的语音增强方法。通过端点检测将语音信号分为语音段与非语音段,对这两种情况的噪声幅度谱分别进行自适应估计,并对谱减法中不具有通用性的假设进行研究从而改进原理公式。实验结果表明,相对于传统谱减法,该方法能更好地抑制音乐噪声,并保持较高清晰度和可懂度,提高了强噪声环境下的语音识别精度和通信质量。     

抗噪音谱估计误差干扰的语音增强处理

在实际应用中通常无法精确估计得到背景噪音谱,传统语音增强效果也随之大大降低。为弱化估计误差引入的干扰,在对数最小均方差估计器（LSA）语音增强方法基础上提出了一套切实可行的增强方案。引入信号检测自动机判别帧成分,针对帧与帧之间的不同特点采取不同级别的噪音抑制处理方案,对确定为噪音帧的部分进行进一步深度抑制,而语音帧部分则沿用改进的LSA方法。实验表明,使用方法能有效抑制背景噪音,特别当噪音谱估计误差较大情况下,相比于LSA该方法具有更优秀的去噪、抗干扰性能。     

一种基于噪声对消与倒谱均值相减的鲁棒语音识别方法

提出一种基于语音增强算法的噪声鲁棒语音识别方法．在语音识别预处理阶段,通过噪声对消语音增强法来抑制噪声提高信噪比．然后对增强语音提取Mel频段倒谱特征参数,并在倒谱域应用倒谱均值相减处理来补偿增强语音中的失真成分和剩余噪声．实验结果表明,在低信噪比（-12—0 dB）条件下,该方法对于数字语音识别具有较好的识别率,其性能明显优于基本的Mel频段倒谱参数识别器、传统的谱减法和噪声对消语音增强法．     

一种强噪音环境中的语音端点检测算法

针对强噪音环境中语音端点检测准确率较低的问题,提出了一种应用在强噪音环境中的语音端点检测算法,结合先验信噪比估计语音增强和改进子带谱熵的算法实现了强噪音中的端点检测。实验结果表明,相比传统端点检测算法,该算法在不同噪声环境下具有较高的鲁棒性,特别是在低信噪比下具有较高的端点检测准确率和较低的误检率。     

结合倒谱阈值频谱估计与瑞利分布模型的语音活动检测

针对传统的似然比语音活动检测的计算语音与噪声统计模型复杂度高,提出结合倒谱阈值估计噪声频谱与瑞利统计模型的语音活动检测方法。该方法先用倒谱阈值估计噪声的频谱,再利用UMPT获得基于瑞利模型的语音判决阈值更新准则。评估了4种不同方法组合的语音活动检测（voice activity detection,VAD）。实验表明：在非平稳噪声环境下该方法的正确检测率优于其它组合的VAD方法。     

A new efficient backward BSS crosstalk-resistant algorithm for automatic blind speech quality enhancement

In last 10 years, several noise reduction (NR) algorithms have been proposed to be combined with the blind source separation techniques to separate speech and noise signals from blind noisy observations. More often, techniques use voice activity detector (VAD) systems for the optimal solution. In this paper, we propose a new backward blind source separation (BBSS) structure that uses the input correlation properties to provide: (i) high convergence rates and good tracking capabilities, since the acoustic environments imply long and time-variant noise paths, and (ii) low misalignment and robustness against different noise type variations and double-talk. The proposed algorithm has an automatic behavior to enhance noisy speech signals, and do not need any VAD systems to separate speech and noise signals. The obtained results in terms of several objective criteria show the good performance properties of the proposed algorithm in comparison with state-of-the-art algorithms.     

An improved noise-robust voice activity detector based on hidden semi-Markov models

To improve the performance of voice activity detector (VAD) in noisy environments, this paper concentrates on three critical aspects related to noise robustness including speech features, feature distributions and temporal dependence. Based on the statistic on TIMIT and NOIZEUS, Mel-frequency cepstrum coefficients (MFCCs) are selected as speech features, Gaussian Mixture distributions (GMD) are applied to associate the observations in MFCC domain with both speech and non-speech states, and Weibull and Gamma distributions are used to explicitly model noise and speech durations, respectively. To integrate these aspects into VAD, the hidden semi-Markov model (HSMM) as a generalized hidden Markov model (HMM) is introduced first. Then the VAD decision is made according to the likelihood ratio test (LRT) incorporating state prior knowledge and modified forward variables of HSMM. We design a recursive way to efficiently calculate modified forward variables. Finally a series of experiments demonstrate: (1) the positive effect of different robustness-related schemes adopted in the proposed VAD; (2) better performance against the standard ITU-T G.729B, Adaptive MultiRate VAD phase 2 (AMR2), Advanced Front-end (AFE), HMM-based VAD and VAD using Laplacian-Gaussian model (LD-GD based VAD).     

基于噪声倒谱阈值频谱估计的语音活动检测

针对低方差频谱估计的语音活动检测(VAD)中Welch频谱估计方法计算量大的问题,提出利用倒谱阈值方法估计VAD中的噪声功率谱.该方法在静音时期为噪声的倒谱设置阈值,利用快速傅里叶变换计算频谱,再更新VAD中的判决阈值.算法复杂度分析与仿真结果表明,该方法的检测性能与Welch方法相当,计算量降低约18%,同时降低整个...     

一种基于EEMD域统计模型的话音激活检测算法

该文提出了一种基于EEMD域统计模型的话音激活检测算法。算法首先利用总体平均经验模态分解(Ensemble Empirical Mode Decomposition,EEMD)对带噪语音进行分解,得到信号的本征模式函数(Intrinsic Mode Function,IMF)分量,选择与原信号的相关性最高的两个分量相加组成主分量;然后对主分量进行频域分解,引入统计模型,求出EEMD域特征参数;最后利用噪声与语音的EEMD域特征参数的不同来进行语音激活检测。实验结果表明,在不同信噪比情况下,本文算法性能优于目前常用的 VAD算法,特别在噪声强度大时体现出明显的优势。     

HMM-Based Gain Modeling for Enhancement of Speech in Noise

Accurate modeling and estimation of speech and noise gains facilitate good performance of speech enhancement methods using data-driven prior models. In this paper, we propose a hidden Markov model (HMM)-based speech enhancement method using explicit gain modeling. Through the introduction of stochastic gain variables, energy variation in both speech and noise is explicitly modeled in a unified framework. The speech gain models the energy variations of the speech phones, typically due to differences in pronunciation and/or different vocalizations of individual speakers. The noise gain helps to improve the tracking of the time-varying energy of nonstationary noise. The expectation-maximization (EM) algorithm is used to perform offline estimation of the time-invariant model parameters. The time-varying model parameters are estimated online using the recursive EM algorithm. The proposed gain modeling techniques are applied to a novel Bayesian speech estimator, and the performance of the proposed enhancement method is evaluated through objective and subjective tests. The experimental results confirm the advantage of explicit gain modeling, particularly for nonstationary noise sources     

Non-linear feature extraction for robust speech recognition in stationary and non-stationary noise

An analysis-based non-linear feature extraction approach is proposed, inspired by a model of how speech amplitude spectra are affected by additive noise. Acoustic features are extracted based on the noise-robust parts of speech spectra without losing discriminative information. Two non-linear processing methods, harmonic demodulation and spectral peak-to-valley ratio locking, are designed to minimize mismatch between clean and noisy speech features. A previously studied method, peak isolation [IEEE Transactions on Speech and Audio Processing 5 (1997) 451], is also discussed with this model. These methods do not require noise estimation and are effective in dealing with both stationary and non-stationary noise. In the presence of additive noise, ASR experiments show that using these techniques in the computation of MFCCs improves recognition performance greatly. For the TI46 isolated digits database, the average recognition rate across several SNRs is improved from 60% (using unmodified MFCCs) to 95% (using the proposed techniques) with additive speech-shaped noise. For the Aurora 2 connected digit-string database, the average recognition rate across different noise types, including non-stationary noise background, and SNRs improves from 58% to 80%.     

基于帧间相关性的语音活动检测方法

为了提高统计模型似然比测试的语音活动检测(VAD)的检测性能,利用前后语音帧间存在的统计相关特性,提出一种改进VAD算法。通过前帧语音频谱分量对先验信噪比进行递归估计,然后利用前一帧的语音检测状态来设计判决阈值,建立了双阈值隐马尔可夫模型语音活动判决规则。实验表明,此帧间相关性VAD算法的检测指标值优于Sohn算法。