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Fast convergence and error free SAF in Multimedia Applications

机译:多媒体应用中的快速收敛性和错误自由

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An Adaptive filter is a filter that self-adjusts its transfer function according to an optimizing algorithm. Because of the complexity of the optimizing algorithms, most adaptive filters are digital filters that perform digital signal processing and adapt their performance based on the input signal. An Adaptive filter is often employed in an environment of unknown Statistics for various purposes such as system identification, inverse modeling for channel equalization, adaptive prediction and interference canceling. Knowing nothing about the environment, the filter is initially set to an arbitrary condition and updated in a step by step manner towards an optimum filter setting. For updating, the least mean-square algorithm is often used for its simplicity and robust performance. However, the L MS algorithm exhibits slow convergence when used with an ill-conditioned input such as speech and requires a high computational cost, especially when the system to identified has a long impulse response. Adaptive Filtering is an important concept in the field of signal processing and has numerous applications in fields such as Multimedia Applications and communications. Examples in speech processing include speech enhancement, echo and interference cancellation and speech coding. Simulations show that the proposed structure converges faster than both an equivalent full band structure at lower computational complexity and recently proposed SAF structures for a colored input.
机译:自适应滤波器是根据优化算法自调节其传递功能的滤波器。由于优化算法的复杂性,大多数自适应滤波器是执行数字信号处理的数字滤波器,并基于输入信号调整它们的性能。自适应滤波器通常在未知统计的环境中用于各种目的,例如系统识别,信道均衡的反向建模,自适应预测和干扰消除。无论是对环境一无知的,滤波器最初被设置为任意条件,并在步骤方式朝向最佳滤波器设置更新。为了更新,最小均方算法通常用于其简单性和强大的性能。然而,当与诸如语音的不良输入使用时,L MS算法表现出缓慢的收敛性,并且需要高计算成本,特别是当系统识别的系统具有长的脉冲响应时。自适应滤波是信号处理领域的重要概念,并且在诸如多媒体应用和通信之类的领域具有许多应用。语音处理中的示例包括语音增强,回声和干扰消除和语音编码。模拟表明,所提出的结构会收敛于较低的计算复杂度的等效满带结构,并且最近提出了用于彩色输入的SAF结构。

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