共查询到19条相似文献,搜索用时 125 毫秒
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一种低信噪比语音的增强算法 总被引:2,自引:0,他引:2
为改善低信噪比环境下语音的质量,论文提出了一种新的语音增强算法。算法首先根据噪声频谱的高斯统计模型得到用先验信噪比形式表示的噪声频谱估计值,然后利用帧内、帧间平滑算法估计每一个频点的先验信噪比,从而能够更好地跟踪先验信噪比的变化。算法接着引入一种简便的估计语音在每一个频点出现概率的方法,得出一种新的语音增强算法。客观测试和非正式听音测试表明:该算法在几乎不损伤语音清晰度的前提下,能够更好地抑制低信噪比语音增强所产生的音乐噪声,同时使语音信噪比得到了明显提高。 相似文献
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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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《电子技术与软件工程》2015,(1)
在低信噪比环境下,为了提高语音端点检测的效果,提出了一种适应于低信噪比环境的语音端点检测方法。基于子带谱熵法,引入正参数对基本的谱熵法进行算法改进,得到改进后的子带谱熵法,通过增加预判环节选择合适的正参数,加大语音信号与噪声信号的区分度,进一步改善在低信噪比环境下算法的效果,得到新的语音端点检测算法。仿真实验表明,新的算法不仅快速高效,具有较强鲁棒性,而且适合在低信噪比环境中较准确的检测出语音端点。 相似文献
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为改善低信噪比环境下语音的质量,论文提出了一种改进相位估计的语音增强算法。算法首先根据语音和噪声频谱的统计模型的对称性得到用先验信噪比倒数形式表示的噪声频谱估计值,然后通过分析低信噪比条件下(0dB)相位估计对于幅度估计的重要性,利用噪声频谱估计值估计每一个频点的相位修正值,并给出了一种优化的先验信噪比估计算法,得到一种新的语音增强算法。由仿真实验给出的客观测试和非正式听音测试表明:该算法处理后取得了较好的效果,在抑制低信噪比语音增强所产生的音乐噪声的前提下,相比未改进相位估计的算法处理后的信号,语音失真度更小,语音质量有明显提高。 相似文献
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语音增强技术在低信噪比情况下,由于语音增强带来的失真使得系统的识别性能严重下降.因此提出一种结合特征空间的倒谱均值归一化算法(CMN)和模型空间的并行模型合并算法(PMC)的语音增强失真补偿技术.实验结果表明,该方法有效提高了低信噪比情况下的语音信号识别率. 相似文献
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通过介绍语音增强的特点,详细分析了最小均方误差对数谱幅度估计(MMSE-LSA)算法,并提出了与MMSELSA算法相匹配的语音激活检测(VAD)算法。该方案计算简单、易于实现且语音增强效果好,能够动态地跟踪背景噪声的变化。最后通过仿真分析,比较了MMSE-LSA与其它几种语音增强算法的增强效果。 相似文献
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We propose a novel phase‐based method for single‐channel speech enhancement to extract and enhance the desired signals in noisy environments by utilizing the phase information. In the method, a phase‐dependent a priori signal‐to‐noise ratio (SNR) is estimated in the log‐mel spectral domain to utilize both the magnitude and phase information of input speech signals. The phase‐dependent estimator is incorporated into the conventional magnitude‐based decision‐directed approach that recursively computes the a priori SNR from noisy speech. Additionally, we reduce the performance degradation owing to the one‐frame delay of the estimated phase‐dependent a priori SNR by using a minimum mean square error (MMSE)‐based and maximum a posteriori (MAP)‐based estimator. In our speech enhancement experiments, the proposed phase‐dependent a priori SNR estimator is shown to improve the output SNR by 2.6 dB for both the MMSE‐based and MAP‐based estimator cases as compared to a conventional magnitude‐based estimator. 相似文献
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谱减法是常用的单通道语音降噪方法,传统谱减法在抑制背景噪声的同时引入了“音乐噪声”,影响听觉效果。为了抑制音乐噪声,提出了一种基于后验信噪比的频域语音增强新方法,当后验信噪比较高时,采用基于后验信噪比的谱减法增强语音信号;当后验信噪比较低时,采用基于后验信噪比的谱衰减方法对含噪语音信号谱线进行衰减,达到语音增强的目的。仿真结果表明,基于后验信噪比的频域语音增强法具有较好的背景噪声和音乐噪声抑制效果,并保持了较好语音可懂度。 相似文献
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Speech enhancement algorithms play an important role in speech signal processing. Over the past several decades, many algorithms have been studied for speech enhancement. A speech enhancement algorithm uses a noise removal method and a statistical model filter to analyze the speech signal in the frequency domain. Spectral subtraction and Wiener filters have been used as representative algorithms. These algorithms have excellent speech enhancement performance, but suffer from deterioration in performance due to specific noise or low signal-to-noise ratio (SNR) environments. In addition, according to estimations of erroneous noise, a noise existing in a voice signal is maintained so that a spectrum corresponding to a voice signal is distorted, or a frame corresponding to a voice signal cannot be retrieved, and voice recognition performance deteriorates. The problem of deterioration in speech recognition performance arises from the difference between speech recognition and training model. We use silence-feature normalization model as a methodology to improve the recognition rate resulting from the difference in the noisy environments. Conventional silence-feature normalization has a problem in that the silent part of the energy increases, which affects recognition performance due to unclear boundaries categorizing the voice. In this study, we use the cepstrum feature of the noise signals in the silence-feature normalization model to improve the performance of silence-feature normalization in a signal with a low SNR by setting a reference value for voiced and unvoiced classification. As a result of recognition rate confirmation, the recognition rates improve in performance, compared with other methods. 相似文献