结合牛顿-拉夫森函数计算语音线谱对参数的高效算法
发布时间：2018-09-03 06:50
【摘要】：提出了计算语音信号线谱对(LSP)参数的高效算法NRSPF。首先利用牛顿-拉夫森函数及斯蒂芬森加速求高阶非线性方程的一个实根,再使用多项式综合除法降阶,最后采用费拉里算法求其余的根,即得LSP参数。通过TI-DSP平台的实例研究表明,NRSPF算法与APF算法相比,迭代次数减少、收敛速度加快,计算量小,并且在精度提高10倍、100倍和1000倍情况下,APF算法可能出现被零除错误和死循环,而NRSPF算法不仅避免了该错误,而且迭代次数增加很少,收敛速度仍然很快,得到更精确的结果。本文提出的算法高效、可靠、实时性强,可应用于超低码率语音实时通信系统、语音编解码器等。
[Abstract]:An efficient algorithm, NRSPF., for calculating the (LSP) parameters of speech signal line spectrum is proposed. First, we use Newton-Raphson function and Stephenson's acceleration to find a real root of higher order nonlinear equation, then reduce the order by polynomial synthesis division method. Finally, we use Ferrari algorithm to find the other roots, that is, LSP parameter. A case study on the TI-DSP platform shows that compared with the APF algorithm, the TI-DSP algorithm has fewer iterations, faster convergence speed and less computation, and it may have zero division errors and dead cycles in the case of 10 times higher accuracy and 1000 times higher accuracy. The NRSPF algorithm not only avoids this error, but also increases the number of iterations very little, and the convergence rate is still fast, and the more accurate results are obtained. The algorithm proposed in this paper is efficient, reliable and real-time. It can be used in ultra-low bit-rate speech real-time communication system, speech codec and so on.
【作者单位】： 苏州大学电子信息学院;
【基金】：国家自然科学基金(61271360,61201213) 苏州市应用基础研究计划(SYG201230)资助
【分类号】：TN912.3
，
本文编号：2219227
[Abstract]:An efficient algorithm, NRSPF., for calculating the (LSP) parameters of speech signal line spectrum is proposed. First, we use Newton-Raphson function and Stephenson's acceleration to find a real root of higher order nonlinear equation, then reduce the order by polynomial synthesis division method. Finally, we use Ferrari algorithm to find the other roots, that is, LSP parameter. A case study on the TI-DSP platform shows that compared with the APF algorithm, the TI-DSP algorithm has fewer iterations, faster convergence speed and less computation, and it may have zero division errors and dead cycles in the case of 10 times higher accuracy and 1000 times higher accuracy. The NRSPF algorithm not only avoids this error, but also increases the number of iterations very little, and the convergence rate is still fast, and the more accurate results are obtained. The algorithm proposed in this paper is efficient, reliable and real-time. It can be used in ultra-low bit-rate speech real-time communication system, speech codec and so on.
【作者单位】： 苏州大学电子信息学院;
【基金】：国家自然科学基金(61271360,61201213) 苏州市应用基础研究计划(SYG201230)资助
【分类号】：TN912.3
，
本文编号：2219227
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