[1]王仕奎,蔡卫平,杨志鸿,等.AMR-WB到AMR转码中合成滤波器转换算法[J].东南大学学报(自然科学版),2010,40(4):676-681.[doi:10.3969/j.issn.1001-0505.2010.04.003]
 Wang Shikui,Cai Weiping,Yang Zhihong,et al.Transformation algorithms of synthesis filter in transcoding from AMR-WB to AMR[J].Journal of Southeast University (Natural Science Edition),2010,40(4):676-681.[doi:10.3969/j.issn.1001-0505.2010.04.003]
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AMR-WB到AMR转码中合成滤波器转换算法()
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《东南大学学报(自然科学版)》[ISSN:1001-0505/CN:32-1178/N]

卷:
40
期数:
2010年第4期
页码:
676-681
栏目:
信息与通信工程
出版日期:
2010-07-20

文章信息/Info

Title:
Transformation algorithms of synthesis filter in transcoding from AMR-WB to AMR
作者:
王仕奎12 蔡卫平1 杨志鸿1 吴镇扬13
1 东南大学信息科学与工程学院, 南京 210096; 2 安徽师范大学物理与电子信息学院,芜湖 241000; 3 东南大学水声信号处理教育部重点实验室(B类筹),南京 210096
Author(s):
Wang Shikui12 Cai Weiping1 Yang Zhihong1 Wu Zhenyang13
1 School of Information Science and Engineering, Southeast University, Nanjing 210096, China
2 College of Physics and Electronic Information, Anhui Normal Univesity, Wuhu 241000, China
3 Key Laboratory of Underwater Acoustic Signal Processing of Ministry of Education, Southeast University, Nanjing 210096, China
关键词:
转码 合成滤波器 Prony算法 自相关 三次样条内插
Keywords:
transcoding synthesis filter Prony algorithm autocorrelation cubic spline interpolation
分类号:
TN912.3
DOI:
10.3969/j.issn.1001-0505.2010.04.003
摘要:
提出AMR-WB到AMR转码中的2种合成滤波器转换算法.第1种是基于采样率转换和Prony算法的转换,首先将AMR-WB合成滤波器的单位采样响应进行采样率转换,然后根据最小二乘法,使得新的滤波器的单位采样响应和采样率转换后的响应的误差最小化.第2种是基于自相关值内插的转换算法,首先由AMR-WB语音的 LPC参数倒推出自相关,然后采用三次样条内插出AMR语音的自相关,最后利用Levinson-Durbin算法计算LPC参数,即得到解码端的合成滤波器.算法复杂度分析表明,2种算法的计算复杂度都低于Tandem转码.实验结果表明,2种算法都可以得到比较小的谱失真.第2种算法的谱失真在浊音帧比第1种算法略大,在清音帧谱失真有时较大,但是由于清音激励的随机性,对合成清音质量影响不大.
Abstract:
Two translation algorithms of synthesis filter are presented in the transcoding from adaptive multi-rate wideband(AMR-WB)to adaptive multi-rate(AMR). The first one is based on the conversion of sampling rate and Prony algorithm: the sampling rate of the unit sampling response of AMR-WB synthesis filter is converted, and the error between the unit sampling response of the translated synthesis filter and the synthesis filter whose sampling rate has been converted is minimized according to the least square method. The second one is based on the interpolation of autocorrelations: the autocorrelations are deduced from the LPC parameters of AMR-WB; the autocorrelations of AMR speech are obtained through cubic spline interpolation; finally, the LPC parameters of AMR are computed through Levinson-Durbin algorithm. Complexity analysis indicates that, compared to tandem transcoding, the computational complexity of these two algorithms is lower. Experimental results show that, the spectral distortion(SD)of these two algorithms is small, but for voiced frames, it is a little larger in the second algorithm than that in the first algorithm. For unvoiced frames, the SD in the second algorithm is sometimes high, but it has little effect on the synthesized speech due to the randomicity of excitation in unvoiced speech.

参考文献/References:

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备注/Memo

备注/Memo:
作者简介: 王仕奎(1970—),男,博士生,讲师; 吴镇扬(联系人),男,教授,博士生导师,zhenyang@seu.edu.cn.
基金项目: 国家自然科学基金资助项目(60971098).
引文格式: 王仕奎,蔡卫平,杨志鸿,等.AMR-WB到AMR转码中合成滤波器转换算法[J].东南大学学报:自然科学版,2010,40(4):676-681. [doi:10.3969/j.issn.1001-0505.2010.04.003]
更新日期/Last Update: 2010-07-20