共查询到19条相似文献,搜索用时 250 毫秒
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针对语音信号在传输和处理过程中不同程度地受周围环境噪声污染的问题,提出一种基于小波变换的改进型语音除噪算法.传统的小波语音除噪算法把信号的高频部分置零,会造成除噪后信号的失真.这里的算法,先对语音信号进行清、浊音分离,然后分别对清音和浊音部分进行不同的阈值处理,不但保留了语音中的高频信息,同时也提高了语音信息的逼真度和信噪比.仿真结果表明,与传统的小波语音除噪算法相比,该算法对含噪语音在高频部分和低频部分都具有很好的去噪效果. 相似文献
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一种改进的子波域语音增强方法 总被引:13,自引:0,他引:13
本文基于文献[3]中提出的子波域去噪技术,并针对语音信号的特点提出了一种改进的子波域语音增强方法。该方法采用软限幅函数对语音信号的子波变换系数作阈值处理以达到去噪的目的。同时,为了防止在抑制噪声的过程中对语音的清音段信息造成损失,首先对语音信号进行了清浊音判别,然后针对不同的判别结果对清音段语音和浊音段语音采用不同的阈值处理方法。仿真实验表明,该方法效果良好且简便易行,是一种有效的语音增强技术。 相似文献
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基于MATLAB的语音增强系统的设计 总被引:1,自引:0,他引:1
语音增强是信号处理领域中的一个重要的组成部分。在许多语音处理的应用中,例如移动通信,语音识别和助听器,语音信号的处理不得不在具有噪声的环境下进行。在过去的几十年里,人们提出了许多方法去消除噪声和减少语音失真,例如谱减法,基于小波的方法,隐式马尔科夫模型法和信号子空间法等。小波分析由于能同时在时域和频域中对信号进行分析,所以它能有效地实现对信号的去噪。介绍了一种语音增强系统的设计方法,采用Least Mean Square(LMS)算法和小波变换相结合的方法对带噪语音进行去噪,并在MATLAB的Simulink环境下建立了该系统的模型。通过对该模型的仿真表明:该方法去噪效果明显,为该系统在硬件上的实现打下了理论基础。 相似文献
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基于小波变换的光寻址电位传感器信号去噪研究 总被引:2,自引:2,他引:0
基于(LAPS)(光寻址电位传感器)技术的生化传感器中的光生电流是一种微弱的非平稳信号,信噪比(SNR)低。为了提取清晰的LAPS信号,且鉴于传统的傅里叶方法去噪后信号失真严重,本文采用小波变换的方法对LAPS信号进行去噪处理。通过小波变换将信号分解为3层,得到各层的小波系数以及阈值。根据每一层系数特点,按阈值进行分别处理,得到新的小波系数。最后根据新的小波系数,重构信号。对去噪后的信号进行频谱分析发现,信号频谱为有效的LAPS信号谱段。将傅里叶去噪和小波去噪方法进行对比发现,小波去噪得到信号的SNR和平滑度(SR)要高于傅里叶去噪,表明小波变换是LAPS信号去噪的有效方法。 相似文献
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语音增强目的是从带噪语音中尽可能纯净的原始语音,即消除含噪语音信号中的噪声成份,提高输入信号的信噪比.在实际应用环境中,语音都会不同程度受到噪声的干扰,噪声会影响语音质量,严重的情况下将语音完全淹没到噪声中,无法分辨.本文将读入的语音信号加入正态随机噪声,然后对含噪声的语音信号进行小波分解,估计噪声的方差,然后获取去噪的阈值并对小波分解的高频系数进行阈值量化,得到去噪后的语音信号.仿真证明此方法具有很好的增强效果. 相似文献
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语音信号与随机噪声在不同尺度上进行小波变换时,其小波变换系数和尺度大小的特性关系存在着不同的特征表现,而且,浊音和清音也各有其特性。给出了一种基于小波变换的维纳滤波语音增强方法;采用维纳滤波对浊音和清音信号的小波变换系数进行不同的处理,既抑制了噪声,又减少了语音段信息的损失,提高了信噪比。仿真结果说明,这是一种有效的语音增强方法。 相似文献
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针对现有深度神经网络语音增强方法对带噪语音的去噪能力有限、语音质量提升不高的问题,提出了一种基于奇异谱分析的深度神经网络语音增强方法。通过引入奇异谱分析算法对带噪语音进行预处理,以初步分离得到语音信号与噪声。接着将语音信号与噪声用于深度神经网络模型得训练,以得到性能更优的网络模型,从而使得本文方法具有更好的性能。最后在重建干净语音的环节中,同时使用神经网络估计得到的对数功率谱和带噪语音的对数功率谱,并加入了权重系数,使得本文提出的方法可以适应不同信噪比的情形,有效的去除背景噪声,降低语音信号的失真。本文通过仿真实验验证了该方法的有效性和鲁棒性。 相似文献
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Telephone channels restrict the bandwidth of speech signals to approximately 0.3-3.3 kHz, with the consequence that the intelligibility of unvoiced sounds may be significantly impaired. To prevent this band limitation of unvoiced sounds while still confining the speech to the telephonic bandwidth, we propose a scheme which, on recognizing the presence of unvoiced sounds extending to 7.6 kHz, frequency maps them into the band 0.3-3.3 kHz. Four mapping laws are considered and the unvoiced speech is compressed using each law. Frequency demapping is employed, and the law that has the best spectral match to the speech spectrum is selected. Voiced speech is band limited from 0.3 to 3.3 kHz. Results measured over 16 ms, a phoneme, and word durations indicate that the adaptive frequency mapping algorithm significantly enhances the recovered speech compared to telephonic speech. Informal listening experiences support these findings. 相似文献
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本文研究图频域内的多通道语音增强,利用图信号处理理论(GSP)构建一种时间-空间维度的联合图拓扑结构,在此基础上设计增强算法进行多通道语音消噪。具体而言,基于输入阵列某个麦克风输入帧间语音顶点信号的时间相关关系,构造时间维度上的一种图拓扑结构;同时针对多通道含噪语音,根据各通道接收信号的空间相关关系,构造空间维度上的一种图拓扑结构。基于时间和空间二种图拓扑构成的联合图拓扑结构,采用图频域内的最小方差无失真响应(MVDR)增强算法,进行多通道语音增强。仿真实验结果表明,在平均客观语音质量评估(PESQ)得分和平均拓展短时客观可懂度(ESTOI)评价指标下,本文所提出的基于联合图拓扑结构的MVDR波束形成(JG-MVDR)方法都优于常规图MVDR波束形成(GMVDR)方法和基于复高斯混合模型的MVDR波束形成(CGMM-MVDR)方法。 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
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基音检测是河南方言语音信号处理中的一个重要环节,针对低信噪比环境下的河南方言语音基音检测准确率低的问题,提出了一种语音信号增强和基音检测相结合的算法.通过多窗谱估计的改进谱减法对语音信号进行降噪处理,对增强后的语音信号用中心削波法消除偏离基音轨迹的野点,再通过自相关法实现基音检测.仿真结果表明,对于低信噪比环境下河南方言语音信号的基音估值检测结果准确,估算出的基音频率和实际基音频率能很好的重合. 相似文献
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在现实生活中,有很多音频信号是带有噪声的.由于噪声的影响,人们听到的音频信号不够清晰.为了提高音频信号的清晰度,需要将有用信号提取出来,从而达到去噪的目的.提供了一种音频去噪的模式:用遗传算法对当前最强干扰噪声进行频率估计,然后用陷波来进行去噪.在客观测试中,ODG值提高了0.469,说明去噪后语音与原始无噪语音更加接... 相似文献
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Statistical-model-based speech enhancement systems 总被引:3,自引:0,他引:3
Ephraim Y. 《Proceedings of the IEEE. Institute of Electrical and Electronics Engineers》1992,80(10):1526-1555
Since the statistics of the speech signal as well as of the noise are not explicitly available, and the most perceptually meaningful distortion measure is not known, model-based approaches have recently been extensively studied and applied to the three basic problems of speech enhancement: signal estimation from a given sample function of noisy speech, signal coding when only noisy speech is available, and recognition of noisy speech signals in man-machine communication. Research on the model-based approach is integrated and put into perspective with other more traditional approaches for speech enhancement. A unified statistical approach for the three basic problems of speech enhancement is developed, using composite source models for the signal and noise and a fairly large set of distortion measures 相似文献
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A hybrid pitch detector characterised by parallel analysis of the speech signal in temporal, spectral and cepstral domains is proposed. The voiced/unvoiced decision and pitch period evaluation is realised by a logical analysis of the results from three domains. The experimental analysis shows the robustness of the detector for noisy and telephone speech.<> 相似文献
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