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针对现有语音增强算法面临残留噪声这一问题,结合人耳听觉系统的掩蔽特性,本文提出了一种优化的语音增强算法。算法分为两级,第一级利用MMSE-LSA谱估计法对带噪语音进行降噪处理,经过处理后,带噪语音信号的信噪比得到了提高。然后,针对第一级增强语音信号中的残余噪声利用人耳听觉掩蔽特性掩蔽掉。为此,算法结合人耳听觉掩蔽特性设计了感知增强滤波器,该滤波器能够有效去除第一级增强语音信号中的残留噪声。仿真实验表明,在各种复杂背景噪声以及信噪比环境下,经过本文算法处理后的增强语音信号残留噪声明显减少,算法提升了增强语音的主观感知质量。 相似文献
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提出了一种基于最小统计和人耳掩蔽特性的语音增强算法,通过最优平滑和最小约束递归平均从含噪语音中估计噪声的均值,推导出一种新的基于掩蔽特性的谱减系数计算公式。实验结果表明,该算法优于传统的掩蔽特性算法,含噪语音经过增强后,残留的音乐噪声更小。 相似文献
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一种基于听觉掩蔽模型的语音增强算法 总被引:14,自引:0,他引:14
本文提出一种基于听觉掩蔽模型的语音增强算法。该算法对应用于语音编码中的听觉掩蔽模型进行了适当的修正,动态地确定第一帧语音信号各个关键频率段的听觉掩蔽阈值,有选择性地进行谱减。计算机仿真表明所提算法优于基本谱减法,不仅信噪比有较大的提高而且有效地减少了主观听觉的失真和残留音乐噪声。 相似文献
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针对在低信噪比条件下难以实现语音端点检测,提出了基于混沌理论的解决方法,采用Duffing方程的间歇混沌特性对语音信号进行检测,同时对谱减法作了改进,根据入耳听觉掩蔽效应的语音增强算法,动态修正谱减系数,有针对性地进行谱减,有效克服了音乐噪声.在信噪比较低的情况下,按照该方案处理后的语音不仅信噪比显著提高,主观听觉失真有效减小,且语音清晰度、可懂度和舒适度极大改善. 相似文献
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结合人耳听觉掩蔽效应,提出一种基于听觉感知加权的卡尔曼滤波语音增强方法。由于人耳对语音的感知主要是通过语音信号频谱分量幅度获得的,引入听觉感知加权滤波器在频域上使共振峰区域残留噪声更多,而共振峰之间及语音幅度谱较低的区域残留噪声减少,这样符合人耳的听觉特性,从而使得主观感觉到的噪声最小。采用语音质量感知评估对语音增强的效果进行评测,与传统的卡尔曼滤波语音增强算法相比,实验结果显示该算法提高了增强语音的质量。 相似文献
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This paper presents new wideband speech coding and integrated speech coding-enhancement systems based on frame-synchronized fast wavelet packet transform algorithms. It also formulates temporal and spectral psychoacoustic models of masking adapted to wavelet packet analysis. The algorithm of the proposed FFT-like overlapped block orthogonal wavelet packet transform permits us to efficiently approximate the auditory critical band decomposition in the time and frequency domains. This allows us to make use of the temporal and spectral masking properties of the human auditory system to decrease the average bit rate of the encoder while perceptually hiding the quantization error. The same wavelet packet representation is used to merge speech enhancement and coding in the context of auditory modeling. The advantage of the method presented in this paper over previous approaches is that perceptual enhancement and coding, which is usually implemented as a cascade of two separate systems, are combined. This leads to a decreased computational load. Experiments show that the proposed wideband coding procedure by itself can achieve transparent coding of speech signals sampled at 16 kHz at an average bit rate of 39.4 kbit/s. The combined speech coding-enhancement procedure achieves higher bit rate values that depend on the residual noise characteristics at the output of the enhancement process 相似文献
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基于时频阈值的小波包语音增强算法 总被引:2,自引:0,他引:2
该文考虑小波域应用语音降噪中听觉掩蔽效应,提出了一种基于时频阈值的小波包语音增强算法。新算法首先通过频域增强方法得到语音粗估计,通过跟踪估计语音时频特性的细节变化,及时调节降噪阈值,然后利用时频阈值对小波包系数进行处理,以达到语音降噪的目的。实验表明,较传统小波域语音降噪方法,新算法在抑制平稳白噪声的同时减小了语音信息的损失,其增强语音的MOS(Mean Opinion Score)评分、输出信噪比、MBSD(Modified Bark Spectral Distortion)测度性能均有明显提高。 相似文献
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针对传统谱减语音增强算法增强后的语音信号会残留明显的"音乐噪声"的问题,采用多频带谱减算法对其进行改进。改进算法的原理是将带噪的语音信号按照频率划分成不同的频带,并使这些频带之间互不交叠,根据频带内带有噪声的语音信号和噪声信号信噪比,利用自适应算法求得该频带的过减因子。仿真结果表明:改进多频带谱减算法的语音增强效果优于传统谱减法。 相似文献
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Electrolarynx (EL) speech provides a valuable means of verbal communication for the laryngectomees. Yet EL speech tends to be less intelligible speech due to the presence of background noise. This paper addresses the issue of EL speech enhancement. The proposed approach takes into account the frequency-domain masking properties of the human auditory system for a subtractive-type enhancement process. Subtractive-type algorithms can efficiently reduce the radiated noise of EL speech but not to reduce the additive noise from the environment due to the use of fixed subtraction parameters. Considering the particular characteristics of EL speech, a new computationally efficient algorithm based on the perceptual weighting technique is developed to adapt the subtraction parameters. This leads to a significant reduction of the unnatural structure of the residual noise. Acoustic and perceptual experiments confirm that the enhanced EL speech is more pleasant to human listeners and the proposed algorithm results in improved performance over classical subtractive-type algorithms. 相似文献
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基于听觉感知的LSA-MMSE改进型语音增强方法 总被引:3,自引:0,他引:3
传统增强方法的增益函数对每个频点都进行估计,必然会引进相对较多的语音失真.为了提高低信噪比下的语音增强效果,提出了一种计算掩蔽概率的方法,得到优化的语音增强方法.基于听觉感知特性,对噪声被掩蔽部分的带噪语音谱和未掩蔽部分采用不同处理方法.增强后的语音可以表示为这两个状态下单独估计的加权和,其中权重与噪声被掩蔽概率有关.通过与Virag的方法、LSA-MMSE估计等方法进行比较,实验结果表明所提的增强方法能在低信噪比下有效地抑制残留噪声的同时保持更小的语音失真. 相似文献
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采用了一种基于人耳听觉掩蔽效应的语音增强算法。该算法通过计算每一帧语音信号各个关键频率段的听觉掩蔽阈值,动态地调整谱减系数,有选择性地进行谱减。通过对采集的坦克舱内含强噪声的语音信号的计算机仿真表明,该算法优于基本谱减法,不仅信噪比有较大的提高而且有效地减少了主观听觉的失真和残留音乐噪声。 相似文献
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基于鲁棒主成分分析(RPCA)的单通道语音增强算法是高斯白噪声环境下语音增强的一种重要处理手段,但其对低秩语音分量处理效果欠佳且无法较好地抑制色噪声。针对此问题,该文提出一种基于白化频谱重排RPCA的改进语音增强算法(WSRRPCA),通过优化噪声白化模型,将色噪声语音增强转换成白噪声语音信号处理,利用频谱重排改进RPCA语音增强处理算法,从而获得色噪声环境下语音信号处理性能的整体提升。仿真实验表明,该算法能够较好地实现色噪声环境下的语音增强,且相对于其他算法具有更佳的噪声抑制和语音质量提升能力。 相似文献
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Two‐Microphone Generalized Sidelobe Canceller with Post‐Filter Based Speech Enhancement in Composite Noise
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This paper describes an algorithm to suppress composite noise in a two‐microphone speech enhancement system for robust hands‐free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal‐dominant time‐frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech‐dominant TFBs are identified among the previously detected nonstationary signal‐dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin‐wise output signal‐to‐noise ratio is obtained with these power estimates and a Wiener post‐filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post‐filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality. 相似文献