改进相位补偿结合谐波重构的语音增强方法

为解决传统单声道语音增强方法在对相位处理时存在的不足以及降噪过程中普遍存在的语音失真问题,提出改进相位补偿结合谐波重构的语音增强方法.通过深度学习模型估计先验信噪比并利用先验信噪比对传统相位谱补偿(PSC)函数进行改进,针对在降噪过程中出现的语音失真问题,对增强后的语音通过谐波重构进行二次增强.实验结果表明,改进相位补...     

基于改进谐波恢复算法的语音增强方法

当信噪比较低时,语音信号的高次谐波部分会完全淹没在噪音中。针对该情况,提出一种基于改进谐波恢复算法的语音增强方法。对经过MMSE-LSA算法语音增强处理后的时域输出语音信号进行非线性处理,得到准周期冲激信号,并将其与原增强信号相乘,突出语音的谐波分量。实验结果表明,改进算法较好地解决了低信噪比时谐波失真的问题,相比传统谐波恢复算法能更好地改善语音高次谐波的质量。     

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。     

运用MMSE先验信噪比估计进行语音增强

针对DD（Decision-Directed）先验信噪比估计方法在处理语音时产生延迟以及非因果先验信噪比估计算法不具实时性的缺点,提出一种MMSE（Minimum Mean Square Error）先验信噪比估计方法。它在高斯语音模型假设的基础上,运用最小均方误差准则直接从带噪信号中估计先验信噪比。通过对增强语音信噪比、Itakura-Saito失真测度以及信号时域图和语谱图仿真,结果表明,该算法比DD算法能更好地抑制“音乐噪声”和防止语音畸变,且相对于非因果先验信噪比估计算法具有更强实时性。     

新型几何谱减语音增强方法

针对几何谱减算法在处理快速变化的语音时产生语音畸变的缺点,提出一种基于最小均方误差算法估计每帧语音信号的每一个频率分量上的平滑系数,产生自适应帧频率分量平滑系数代替固定值的平滑系数来估计先验信噪比,从而得到更加接近于真实情况的先验信噪比。通过计算板仓-斋藤距离,及利用仿真波形图、语谱图对算法进行客观测试,结果表明新算法相对其他谱减法在相同的去噪度下,语音畸变度最小且几乎察觉不到音乐噪声;特别是在低信噪比非平稳环境下,相对其他谱减法的优势更加显著。     

一种新的强噪声环境下的语音增强算法

针对强噪声环境下语音增强中噪声估计和先验信噪比估计算法导致的语音失真和音乐噪声的问题,利用语音和噪声的统计模型的对称性得到一种噪声幅度的估计值为参考,提出了一种噪声估计算法,改进了先验信噪比估计算法,形成了一种新的增强算法,适用于强噪声环境下的语音增强。由仿真实验给出的客观评分看出,在0 dB乃至－5 dB条件下,给出信噪比估计算法能够有效减小信号失真,基本上没有残留音乐噪声。     

基于增益字典查询的语音增强算法

对于基于统计模型的语音增强算法,不同分布模型对应于不同的增益函数,由于语音信号的不确定性,没有一种分布函数能准确对语音和噪声谱的分布建模,因此任何一种固定的统计模型均会存在一定的误差。所以提出一种增益字典查询的语音增强算法,该算法通过采用对数谱失真准则对一个语音噪声库进行增益的训练,得到一个增益的字典,其中输入为先验信噪比和后验信噪比的估计值。最后采用ITU-T P.826 PESQ、分段信噪比、总信噪比和对数谱失真对该算法进行了测试,并与基于高斯分布模型、拉普拉斯分布模型的算法进行了对比。实验结果表明,该算法无论在非平稳噪声还是平稳噪声环境下都比其他几种算法增强效果好,且音乐噪声和残留背景噪声也可以得到很好的抑制。     

稀疏低秩模型及相位谱补偿的语音增强算法

针对现有的语音增强算法存在增强效果差、语音信号失真等问题,提出了稀疏低秩模型及改进型相位谱补偿的语音增强算法。首先,用稀疏低秩模型处理含噪语音的幅度谱,得到分离后的语音。接着,用归一化最小均方自适应滤波算法优化相位谱补偿算法的补偿因子。然后,对稀疏低秩分离后的语音进行改进型相位谱补偿处理,得到最终增强的语音。最后,对增强后的语音进行感知语音质量评价分析及频谱分析。实验结果表明,该方法能够有效地去除噪声,并且在低信噪比的情况下,可以保持语音的清晰度。     

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。     

改进增益函数的MMSE语音增强算法

讨论了基于语音短时对数谱最小均方误差(MMSE-STSA)的语音增强算法,将先验信噪比估计引入增益函数的计算中,有效消除噪声.在带噪信号模型中引入语音存在的不确定度,估计出每个频点的先验无声概率,对增益函数进行改进.通过客观与主观两种评价方法将改进算法与小波变换算法和MMSE估计算法进行比较,实验结果表明,改进算法能更好地抑制背景噪声并且使增强后的语音有较小的失真,增加语音清晰度和理解度.     

Speech enhancement based on undecimated wavelet packet-perceptual filterbanks and MMSE–STSA estimation in various noise environments

In this paper, we proposed a new speech enhancement system, which integrates a perceptual filterbank and minimum mean square error–short time spectral amplitude (MMSE–STSA) estimation, modified according to speech presence uncertainty. The perceptual filterbank was designed by adjusting undecimated wavelet packet decomposition (UWPD) tree, according to critical bands of psycho-acoustic model of human auditory system. The MMSE–STSA estimation (modified according to speech presence uncertainty) was used for estimation of speech in undecimated wavelet packet domain. The perceptual filterbank provides a good auditory representation (sufficient frequency resolution), good perceptual quality of speech and low computational load. The MMSE–STSA estimator is based on a priori SNR estimation. A priori SNR estimation, which is a key parameter in MMSE–STSA estimator, was performed by using “decision directed method.” The “decision directed method” provides a trade off between noise reduction and signal distortion when correctly tuned. The experiments were conducted for various noise types. The results of proposed method were compared with those of other popular methods, Wiener estimation and MMSE–log spectral amplitude (MMSE–LSA) estimation in frequency domain. To test the performance of the proposed speech enhancement system, three objective quality measurement tests (SNR, segSNR and Itakura–Saito distance (ISd)) were conducted for various noise types and SNRs. Experimental results and objective quality measurement test results proved the performance of proposed speech enhancement system. The proposed speech enhancement system provided sufficient noise reduction and good intelligibility and perceptual quality, without causing considerable signal distortion and musical background noise.     

Simultaneous Detection and Estimation Approach for Speech Enhancement

In this paper, we present a simultaneous detection and estimation approach for speech enhancement. A detector for speech presence in the short-time Fourier transform domain is combined with an estimator, which jointly minimizes a cost function that takes into account both detection and estimation errors. Cost parameters control the tradeoff between speech distortion, caused by missed detection of speech components and residual musical noise resulting from false-detection. Furthermore, a modified decision-directed a priori signal-to-noise ratio (SNR) estimation is proposed for transient-noise environments. Experimental results demonstrate the advantage of using the proposed simultaneous detection and estimation approach with the proposed a priori SNR estimator, which facilitate suppression of transient noise with a controlled level of speech distortion.     

基于先验信噪比估计和增益平滑的语音增强

针对语音增强算法残留“音乐噪声”的问题,分析了基于先验信噪比估计的语音增强算法,并在此基础上提出自适应先验信噪比估计与增益平滑相结合的方法.这种方法先对先验信嗓比进行估计,然后对增益函数进行平滑,减小相邻增益函数的随机跳变,弥补了传统先验信噪比估计的不足.最后对含高斯白噪声的语音信号进行处理,仿真结果表明,该算法在抑制“音乐噪声”的效果上得到一定改善,提高了语音增强的性能.     

A gain factor adapted by masking property and SNR variation for speech enhancement in colored-noise corruptions

A gain factor adapted by both the intra-frame masking properties of the human auditory system and the inter-frame SNR variation is proposed to enhance a speech signal corrupted by additive noise. In this article we employ an averaging factor, varying with time–frequency, to improve the estimate of the a priori SNR. In turn, this SNR estimate is utilized to adapt a gain factor for speech enhancement. This gain factor reduces the spectral variation over successive frames, so the effect of musical residual noise is mitigated. In addition, the simultaneous masking property of the human ears is also employed to adapt the gain factor. Imperceptive residual noise with energy below the noise masking threshold is retained, resulting in a reduction of speech distortion. Experimental results show that the proposed scheme can efficiently reduce the effect of musical residual noise.     

Improved Signal-to-Noise Ratio Estimation for Speech Enhancement

This paper addresses the problem of single-microphone speech enhancement in noisy environments. State-of-the-art short-time noise reduction techniques are most often expressed as a spectral gain depending on the signal-to-noise ratio (SNR). The well-known decision-directed (DD) approach drastically limits the level of musical noise, but the estimated a priori SNR is biased since it depends on the speech spectrum estimation in the previous frame. Therefore, the gain function matches the previous frame rather than the current one which degrades the noise reduction performance. The consequence of this bias is an annoying reverberation effect. We propose a method called two-step noise reduction (TSNR) technique which solves this problem while maintaining the benefits of the decision-directed approach. The estimation of the a priori SNR is refined by a second step to remove the bias of the DD approach, thus removing the reverberation effect. However, classic short-time noise reduction techniques, including TSNR, introduce harmonic distortion in enhanced speech because of the unreliability of estimators for small signal-to-noise ratios. This is mainly due to the difficult task of noise power spectrum density (PSD) estimation in single-microphone schemes. To overcome this problem, we propose a method called harmonic regeneration noise reduction (HRNR). A nonlinearity is used to regenerate the degraded harmonics of the distorted signal in an efficient way. The resulting artificial signal is produced in order to refine the a priori SNR used to compute a spectral gain able to preserve the speech harmonics. These methods are analyzed and objective and formal subjective test results between HRNR and TSNR techniques are provided. A significant improvement is brought by HRNR compared to TSNR thanks to the preservation of harmonics.