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哪位大侠能帮我把这段话翻译成英文?万分感谢~~~
在语音信号的理论和应用中,所用的语音数据大部分都是在接近理想的条件下采集的。然而,在语音通信过程中会不可避免的受到各种噪声的干扰,这种干扰将最终导致接收者接收到的语音信号...
在语音信号的理论和应用中,所用的语音数据大部分都是在接近理想的条件下采集的。然而,在语音通信过程中会不可避免的受到各种噪声的干扰,这种干扰将最终导致接收者接收到的语音信号已非纯净的原始语音,而是受到噪声污染后的带噪语音。
语音增强作为一种抑制背景噪声、提高语音质量的有效手段,在实际应用中有重要价值。语音增强是一种当语音通信系统的输入或输出信号受到噪声干扰时提高其性能的技术。它的主要目的是降低背景噪声,提高语音质量或抑制同声道语音干扰。简单的讲,就是从带噪声语音信号中提取尽可能纯净的原始语音。传统的语音增强的方法有很多,如:滤波器法语音增强,非线性处理语音增强,减谱法语音增强,自相关相减法语音增强等等。噪声对消的基本原理是从带噪语音中减去噪声。自适应滤波是研究一类结构和参数可以改变或调整的系统。自适应滤波器通常采用FIR滤波器,其系数可以采用最小均方(LMS)法进行估计。最小均方(LMS)算法的基本思路与梯度下降法是一致的,不同之处仅在于计算中用梯度向量的估计来代替真实的梯度。所以说,LMS算法是由梯度下降法导出的,是对梯度下降法的近似简化,更符合实际应用
本课题研究自适应滤波在语音增强中的应用,用LMS算法设计FIR自适应滤波器,用自适应干扰对消原理滤除噪声,并通过仿真研究语音增强的效果。
时间很急~~~谢谢了~~~感激不尽!
不要用翻译软件的~~~~ 展开
语音增强作为一种抑制背景噪声、提高语音质量的有效手段,在实际应用中有重要价值。语音增强是一种当语音通信系统的输入或输出信号受到噪声干扰时提高其性能的技术。它的主要目的是降低背景噪声,提高语音质量或抑制同声道语音干扰。简单的讲,就是从带噪声语音信号中提取尽可能纯净的原始语音。传统的语音增强的方法有很多,如:滤波器法语音增强,非线性处理语音增强,减谱法语音增强,自相关相减法语音增强等等。噪声对消的基本原理是从带噪语音中减去噪声。自适应滤波是研究一类结构和参数可以改变或调整的系统。自适应滤波器通常采用FIR滤波器,其系数可以采用最小均方(LMS)法进行估计。最小均方(LMS)算法的基本思路与梯度下降法是一致的,不同之处仅在于计算中用梯度向量的估计来代替真实的梯度。所以说,LMS算法是由梯度下降法导出的,是对梯度下降法的近似简化,更符合实际应用
本课题研究自适应滤波在语音增强中的应用,用LMS算法设计FIR自适应滤波器,用自适应干扰对消原理滤除噪声,并通过仿真研究语音增强的效果。
时间很急~~~谢谢了~~~感激不尽!
不要用翻译软件的~~~~ 展开
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In the speech signal theory and application, the use of voice and data are mostly in nearly ideal conditions of the acquisition. However, in voice communications will be inevitable in the process of being subjected to various noise interference, such interference would lead the recipient to receive the signal is not the pure voice of the original voice, but by the noise pollution after the Noisy voice.
Speech enhancement as a background noise suppression, improve voice quality and effective means, in practical applications have significant value. When the speech enhancement is a voice communications system imported or exported by the noise signal to raise the performance of the technology. Its main purpose is to reduce background noise, improve voice quality or inhibit the same voice channel interference. Simply speaking, with the noise from the speech signal to extract as much as possible the original pure voice. Speech enhancement of the traditional methods are many, such as: filter speech enhancement law, non-linear process speech enhancement, by Spectrum speech enhancement, since the relevant phase subtraction speech enhancement, and so on. Noise cancellation is the basic principle of Noisy Speech in less noise. Adaptive filtering is a class structure and parameters can be changed or adjusted in the system. Adaptive filters commonly used FIR filter, its coefficient can be used both to the smallest (LMS) method estimates. Minimum square (LMS) algorithm and the basic idea gradient method is the same, only different in the gradient vector used to calculate the estimated true to replace the gradient. Therefore, LMS algorithm is derived from the gradient method, the gradient of the approximate simplified, more in line with the practical application
The research adaptive filter speech enhancement in the application of LMS algorithm using adaptive FIR filter design, using principles of adaptive interference cancellation filtered noise, and enhanced voice through the simulation of the effect.
Speech enhancement as a background noise suppression, improve voice quality and effective means, in practical applications have significant value. When the speech enhancement is a voice communications system imported or exported by the noise signal to raise the performance of the technology. Its main purpose is to reduce background noise, improve voice quality or inhibit the same voice channel interference. Simply speaking, with the noise from the speech signal to extract as much as possible the original pure voice. Speech enhancement of the traditional methods are many, such as: filter speech enhancement law, non-linear process speech enhancement, by Spectrum speech enhancement, since the relevant phase subtraction speech enhancement, and so on. Noise cancellation is the basic principle of Noisy Speech in less noise. Adaptive filtering is a class structure and parameters can be changed or adjusted in the system. Adaptive filters commonly used FIR filter, its coefficient can be used both to the smallest (LMS) method estimates. Minimum square (LMS) algorithm and the basic idea gradient method is the same, only different in the gradient vector used to calculate the estimated true to replace the gradient. Therefore, LMS algorithm is derived from the gradient method, the gradient of the approximate simplified, more in line with the practical application
The research adaptive filter speech enhancement in the application of LMS algorithm using adaptive FIR filter design, using principles of adaptive interference cancellation filtered noise, and enhanced voice through the simulation of the effect.
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In the pronunciation signal theory and the application, the pronunciation data which uses majority of all is gathers under the close ideal condition.However, can inevitable receive each kind of noise in the pronunciation course of communications the disturbance, this kind of disturbance finally will cause the receive receive pronunciation signal already non-pure primitive pronunciation, after but will be receives the noise pollution the belt chirp pronunciation. The pronunciation enhancement took one kind of suppression background noise, improves the pronunciation quality the effective method, has the important value in the practical application.The pronunciation enhancement is one kind when the pronunciation communications system input or the output signal receive the noise jamming enhances its performance the technology.Its main purpose reduces the background noise, improves the pronunciation quality or suppresses the same sound track pronunciation disturbance.Simple speaking, is withdraws as far as possible the pure primitive pronunciation from the belt noise pronunciation signal.The traditional pronunciation strengthens the method has very much, for example: The filter law pronunciations strengthen, the non-linear processing pronunciation strengthens, reduces the spectral method pronunciation to strengthen, autocorrelation subtractive method pronunciation enhancement and so on.The noise cancels the basic principle is subtracts the noise from the belt chirp pronunciation.The adaptive filtering is studies a kind of structure and the parameter may change or the adjustment system.The auto-adapted filter usually use the FIR filter, its coefficient may use the smallest mean square (LMS) law to carry on the estimate.The smallest mean square (LMS) algorithm
basic mentality with the gradient drop law is consistent, the similarity only lies in the computation to replace the real gradient with the gradient vector estimate.Therefore said, the LMS algorithm is derives by the gradient drop law, is drops to the gradient method approximate simplification, conforms to practical application this topic to study the adaptive filtering to strengthen in the pronunciation the application, designs the FIR auto-adapted filter with the LMS algorithm, with the auto-adapted disturbance cancellation principle filtration noise, and the effect which strengthens through the simulation research pronunciation
basic mentality with the gradient drop law is consistent, the similarity only lies in the computation to replace the real gradient with the gradient vector estimate.Therefore said, the LMS algorithm is derives by the gradient drop law, is drops to the gradient method approximate simplification, conforms to practical application this topic to study the adaptive filtering to strengthen in the pronunciation the application, designs the FIR auto-adapted filter with the LMS algorithm, with the auto-adapted disturbance cancellation principle filtration noise, and the effect which strengthens through the simulation research pronunciation
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In the speech signal theory and application, the use of voice and data are mostly in nearly ideal conditions of the acquisition. However, in voice communications will be inevitable in the process of being subjected to various noise interference, such interference would lead the recipient to receive the signal is not the pure voice of the original voice, but by the noise pollution after the Noisy voice.
Speech enhancement as a background noise suppression, improve voice quality and effective means, in practical applications have significant value. When the speech enhancement is a voice communications system imported or exported by the noise signal to raise the performance of the technology. Its main purpose is to reduce background noise, improve voice quality or inhibit the same voice channel interference. Simply speaking, with the noise from the speech signal to extract as much as possible the original pure voice. Speech enhancement of the traditional methods are many, such as: filter speech enhancement law, non-linear process speech enhancement, by Spectrum speech enhancement, since the relevant phase subtraction speech enhancement, and so on. Noise cancellation is the basic principle of Noisy Speech in less noise. Adaptive filtering is a class structure and parameters can be changed or adjusted in the system. Adaptive filters commonly used FIR filter, its coefficient can be used both to the smallest (LMS) method estimates. Minimum square (LMS) algorithm and the basic idea gradient method is the same, only different in the gradient vector used to calculate the estimated true to replace the gradient. Therefore, LMS algorithm is derived from the gradient method, the gradient of the approximate simplified, more in line with the practical application
The research adaptive filter speech enhancement in the application of LMS algorithm using adaptive FIR filter design, using principles of adaptive interference cancellation filtered noise, and enhanced voice through the simulation of the effect.
Speech enhancement as a background noise suppression, improve voice quality and effective means, in practical applications have significant value. When the speech enhancement is a voice communications system imported or exported by the noise signal to raise the performance of the technology. Its main purpose is to reduce background noise, improve voice quality or inhibit the same voice channel interference. Simply speaking, with the noise from the speech signal to extract as much as possible the original pure voice. Speech enhancement of the traditional methods are many, such as: filter speech enhancement law, non-linear process speech enhancement, by Spectrum speech enhancement, since the relevant phase subtraction speech enhancement, and so on. Noise cancellation is the basic principle of Noisy Speech in less noise. Adaptive filtering is a class structure and parameters can be changed or adjusted in the system. Adaptive filters commonly used FIR filter, its coefficient can be used both to the smallest (LMS) method estimates. Minimum square (LMS) algorithm and the basic idea gradient method is the same, only different in the gradient vector used to calculate the estimated true to replace the gradient. Therefore, LMS algorithm is derived from the gradient method, the gradient of the approximate simplified, more in line with the practical application
The research adaptive filter speech enhancement in the application of LMS algorithm using adaptive FIR filter design, using principles of adaptive interference cancellation filtered noise, and enhanced voice through the simulation of the effect.
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下1个有道桌面词典,打上汉语就行
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