基于熵函数的语音端点检测算法研究

本文在分析基本谱熵端点检测算法局限的基础上,引入基于二阶累积量的门限更新方法,加入短时能量参数,提出基于加权谱熵的检测方法;此外,引入特征空间能量熵定义,建立新的门限确定准则,提出基于特征空间能量熵的检测方法.通过对平稳高斯白噪声、M109坦克噪声和F16战斗机噪声这三种典型噪声环境下信噪比(SNR)从-5dB到20dB的带噪语音信号进行的仿真实验分析表明,所提两种方法能更为准确地检测到语音的端点.     

掩蔽模型对语音增强效果影响的研究

对两种最常用的噪声估计算法--VAD噪声检测估计和基于最小值统计特性的噪声估计法,运用不同的掩蔽模型计算了增强前后的信噪比SNR.结果表明:掩蔽效应的运用对语音增强效果的影响很大,在不使用掩蔽效应的情况下,最优最小统计值法语音增强的SNR增加很多,从而可能引入更大的音乐噪声.但无论对VAD噪声检测估计法和基于最小值统计特性的噪声估计法,掩蔽阔值偏移量的少量变化对SNR的影响都不是很大,其原因主要是语音信号本身的幅度较大.     

一种新的语音和噪声活动检测算法及其在手机双麦克风消噪系统中的应用

针对现有双通道语音活动检测(Voice Activity Detection, VAD)算法依赖于固定阈值难以在多种噪声环境下准确地检测语音和噪声,应用于手机消噪系统会造成语音失真或噪声消除不好等问题,该文提出一种基于神经网络的VAD算法,该算法以分频带能量差和归一化互通道相关为特征,采用神经网络对语音和噪声进行分类。在此基础上,将神经网络VAD与基于互通道信号功率比值的VAD相结合,提出一种新的适用于手机消噪系统的语音和噪声活动检测算法分别对语音和噪声进行检测,并以此进行噪声抑制处理,减少了消噪系统因VAD误判而造成的性能下降。实验结果表明,该处理方法在抑制背景噪声和减少语音失真等方面优于现有的消噪算法,对于方向性语音干扰也有很好的抑制效果。     

一种基于改进的谱减法的语音增强算法

为改善语音质量,提高语音识别系统的性能,提出了一种新的基于改进的谱减法的语音增强算法。新算法在所有噪声都能转化为加性高斯白噪声的基础上,依据高斯白噪声幅度谱服从瑞利分布的特点,对噪声幅度谱进行精确的估计,并采用频带方差进行端点检测以及时检出语音。仿真分析表明：该方法能够较好地抑制噪声,噪声消除效果较传统算法具有明显的提高。     

基于倒谱特征的语音端点检测算法研究

本文在讨论基于倒谱距离语音端点检测算法不足的基础上,提出了两种改进方案.通过对三种典型噪声环境下信噪比(SNR)从-5dB到20dB的带噪语音信号进行的仿真实验结果表明,所提两种改进方案能更为准确地检测到语音的端点.     

基于稀疏性的相位谱补偿语音增强算法

相位谱补偿语音增强算法通过调整相位谱对噪声进行压缩,提高重构信号的质量。针对传统的相位谱补偿(phase spectrum compensation, PSC)语音增强算法采用固定的相位补偿因子,且算法的性能易受噪声估计准确性的影响,提出了一种基于稀疏性的相位谱补偿(sparsity-based phase spectrum compensation, SPSC)语音增强算法。首先,利用噪声估计算法得到噪声幅度谱,利用基于幅度谱的语音增强算法得到目标语音幅度谱;接着,通过噪声和目标语音幅度谱之间的局部信噪比(Signal-to-Noise Ratio, SNR)来估计谱时间稀疏性;然后,利用sigmoid函数改进相位补偿因子,联合补偿因子和谱时间稀疏性,得到SPSC函数。最后,使用SPSC函数对相位谱中的谱分量进行补偿,通过短时傅里叶逆变换得到最终增强后的语音信号。仿真实验表明,在四种不同背景噪声的低信噪比下,新的相位谱补偿算法使增强语音获得了更好的LSD、PESQ和segSNR指标,说明新的算法在低信噪比下,可以有效恢复带噪语音中的语音成分,对噪声抑制效果明显,增强语音的质量和听感均有一定提升。      

基于自适应噪声估计的小波阈值语音增强

文中提出了一种基于小波阈值和自适应噪声估计方法的语音增强算法。该算法直接利用含噪语音信号估计出信噪比SNR,并通过该值调整小波阈值,从而实现了小波阈值的自适应变化。针对噪声的小波变换模值随尺度增大而减小的特性,采用了随尺度变化的小波阈值。并且改进了小波阈值函数。实验数据表明,本文算法在多种噪声环境下,均有较好的语音增强效果。并且在抑制噪声的同时,减少了语音失真。     

噪声功率不确定性区间估计和降低SNR WALL恶化的能量检测算法

能量检测是认知无线电系统中广泛采用的空闲频谱检测方案,但其性能受到噪声功率不确定性(NPU)的严重影响。该文提出一种新颖的复杂度较低的NPU区间估计算法,并且从理论上分析了估计的噪声功率对能量检测信噪比墙(SNR WALL)恶化的影响,得出了SNR WALL恶化性定理。进一步基于门限修正提出一种改进的能量检测算法以消除SNR WALL恶化。仿真结果表明,该算法能较为精确地估计NPU区间,并且验证了SNR WALL恶化性定理的正确性;同时,改进的能量检测算法性能要优于稳健的统计方案(RSA)能量检测的结果,并且改进后降低了SNR WALL恶化,提高了检测的鲁棒性。     

非平稳噪声环境下的噪声估计算法

通过对噪音和语音频谱的分析,针对航空背景噪声的特性,提出一种用于语音增强的新的噪声估计算法。通常的噪声估计一般利用语音端点检测方法,取噪声段的谱平均值作为待估计的噪声谱,但该方法在信噪比较低时性能下降严重。笔者提出的基于频率段能量比的噪音谱估计方法,不依赖于语音端点检测而直接由语音帧来估计噪音谱,通过计算一帧语音中各频率段中能量比,以判断该帧是否含有语音来修正噪声谱估计的计算因子。算法提高了谱减法的适用范围,还在一般谱相减方法的基础上提出了改进的谱相减算法。     

基于多带谱相减的语音端点检测算法

为了提高噪声环境下语音端点检测的鲁棒性，介绍了一种基于多带谱相减的语音端点检测算法．仿真结果表明，与传统的基于短时能量，过零率语音端点检测算法相比，在低信噪比环境下，该算法仍可以有效的检测出起止端点．     

采用子带长时信号变化特征的稳健语音活动检测

为提高语音活动检测(VAD)在低信噪比下的准确率,提出了一种基于子带长时信号变化特征的VAD算法.将语音信号转换到频域,并分解为几个不重复的子频带,对这些子带信号分别提取长时信号变化特征,然后采用GMM在线建立语音和非语音模型,以模型的似然比进行VAD判决.实验结果表明,算法在较低的信噪比下能够显著地提高语音活动检测的准确率,且在多种噪声环境和信噪比条件下具有较好的稳健性.应用于语音识别系统的实验表明,该算法能有效提高噪声环境下的语音识别率.     

Robust Voice Activity Detection Based on Concept of Modulation Transfer Function in Noisy Reverberant Environments

Voice activity detection (VAD) is used to detect speech and non-speech periods from observed speech signals. It is an important front-end technique for many speech technology applications. Many VAD methods have been proposed. However most of them have been applied under clean or noisy conditions. Only a few methods have been proposed for reverberant conditions, particularly under noisy reverberant conditions. We therefore need to understand the ill effects of noise and reverberation on speech to design an accurate and robust method of VAD under noisy reverberant conditions. The ill effects of noise and reverberation for speech can be regarded as the modulation transfer function (MTF) under noisy and reverberant conditions. Therefore, our study is based on the MTF concept to reduce the ill effects of noise and reverberation on speech, and propose a robust VAD method that we obtained in this study. Noise reduction and dereverberation were first applied to the temporal power envelope of the speech signal to restore the temporal power envelope with this method. Then, power thresholding as a VAD decision was designed based on the restored temporal power envelope. A method of estimating the signal to noise ratio (SNR) was proposed to accurately estimate the SNR in the noise reduction stage. Experiments under both artificial and realistic noisy reverberant conditions were carried out to evaluate the performance of the proposed method of VAD and it was compared with conventional VAD methods. The results revealed that the proposed method significantly outperformed the conventional methods under artificial and realistic noisy reverberant conditions.     

基于DCT分带谱熵与信号分解的高精度基音检测算法

本文就低速率WI语音编码中的基音检测技术进行研究,针对基音检测在不同噪声与信噪比下容易发生清浊误判的问题,在基音检测前端引入基于DCT分带谱熵的语音检测算法划分语音段与非语音段;为了向基音检测算法提供更能准确反映基音周期实际变化的输入语音,基于谐波-噪声模型提出了一种改进的DCT域语音分解算法.然后,根据变形的MCAMDF(Modified Circular Average Magnitude Difference Function)与NCCF(Normalized Cross-Correlation Function)的峰值共性,结合上述两项基音检测前端处理技术,提出了MCAMDF-NCCF基音检测组合算法.为了满足不同环境下WI编码器对基音检测高精度的要求,在合成端更准确地恢复相位轨迹,本文又基于MCAMDF-NCCF算法提出了高精度MCAMDF-NCCF-FRAC基音检测算法以计算分数基音.将算法应用与2kb/s WI编码器,主观A/B听力测试结果表明,本文提出的基音检测算法在低信噪比下明显抑制了基音加倍减半及清浊误判现象的发生,得到了优异的基音检测结果,合成语音质量完全满足低速率WI编码器对基音检测技术的要求.     

基于长时信息的自适应话音激活检测

语音信号的长时信息应用于话音激活检测中表现优越.利用三种听觉滤波器组,对语音信号进行非线性的谱分解,本文提出了六种基于听觉滤波器组的长时信息,并提出了基于长时信息的自适应话音激活检测算法.该算法无需训练数据,根据多种长时信息,直接在待测信号中挑选出类别明确的信号,然后利用这些信号训练分类模型,对待测信号按帧进行语音-非语音分类.在TIMIT语音库和NOISEX-92噪声库上的实验表明,该算法在极低信噪比环境下,仍表现出更高的准确性和更强的稳健性.同时,在线实验表明,算法在实时处理中仍能取得优异的性能.     

基于扩展谱相减的RCAF基音周期检测算法

针对传统基音检测算法在信噪比低的情况下提取的基音周期错误率较高,该文提出了一种基于RCAF (Reverse CAMDF Autocorrelation Function)搜索试探平滑的基音轨迹提取方法。采用自适应判决准则的扩展谱相减进行语音增强,在语音段实现了对噪声信号的估计。应用RCAF算法提取基音周期,通过搜索试探平滑算法对提取出的基音周期进行平滑处理。该算法降低了误判率,提高了提取精度。仿真结果表明,该算法在-10dB信噪比情况下,其性能优于传统的CAMDF和AWAC等方法。     

基于MEFF倒谱距离与对数能量融合的语音端点检测方法

为提高低信噪比环境下语音端点检测算法性能不高的问题,提出将MFCC倒谱距离与对数能量结合进行端点检测.首先,对语音计算对数能量,然后计算改进的倒谱距离,将MFCC倒谱距离与对数能量融合,获得了一种新的语音参数,该参数能有效地提高低信噪比情况下语音与噪声的区别,对参数进行顺利滤波用于语音端点检测,采用自动更新的双阈值进行语音端判别.仿真实验表明,该算法具有较好的适用不同噪声,在低信噪比下依然能获得比较理想的端点检测效果.     

A robust voice activity detector for wireless communications usingsoft computing

Discontinuous transmission based on speech/pause detection represents a valid solution to improve the spectral efficiency of new generation wireless communication systems. In this context, robust voice activity detection (VAD) algorithms are required, as traditional solutions present a high misclassification rate in the presence of the background noise typical of mobile environments. This paper presents a voice detection algorithm which is robust to noisy environments, thanks to a new methodology adopted for the matching process. More specifically, the VAD proposed is based on a pattern recognition approach in which the matching phase is performed by a set of six fuzzy rules, trained by means of a new hybrid learning tool. A series of objective tests performed on a large speech database, varying the signal-to-noise ratio (SNR), the types of background noise, and the input signal level, showed that, as compared with the VAD standardized by ITU-T in Recommendation G.729 annex B, the fuzzy VAD, on average, achieves an improvement in reduction both of the activity factor of about 25% and of the clipping introduced of about 43%. Informal listening tests also confirm an improvement in the perceived speech quality     

基于语谱图的语音端点检测算法

利用语音在语谱图中表现出的不同特征,提出了一种基于语谱图的语音端点检测算法。首先利用基音频率检测的原理在语谱图矩阵中搜索浊音段,然后计算出浊音段的信噪比,再根据信噪比和语谱图矩阵中浊音段的峰值进行完整的端点检测。因多数突发噪声并没有稳定的频率或者频率不在人的基音频率范围内,因此,该算法能够很好地抑制突发噪声的干扰,实验结果表明,在信噪比为10dB以上时该算法能够准确检测出语音的端点位置。     

Sparse Representation with Optimized Learned Dictionary for Robust Voice Activity Detection

Traditionally, most of voice activity detection (VAD) methods are based on speech features such as spectrum, temporal energy, and periodicity. The robustness of these features plays a critical role on the performance of VAD. However, since these features are always directly generated from observed signal, the robustness of these features would be significantly degraded in non-stationary noise environments, especially at low level signal-to-noise ratio (SNR) condition. This paper proposes a kind of robust feature for VAD based on sparse representation with an optimized learned dictionary. To do so, a speech dictionary and a noise dictionary are first learned from speech corpus and noise corpus, respectively. Then an optimization algorithm is designed to reduce the mutual coherence between the two learned dictionaries. After that the proposed feature is generated from the optimized dictionary-based sparse representation, and a VAD method is derived from the proposed feature. The proposed method is evaluated over seven types of noise and four types of SNR level, experimental results show that the optimized dictionary is important for enhancing the robustness of the proposed method, and the proposed method performs well under non-stationary noise, especially at low level SNR condition.