Real time speech compression by using code excited linear prediction algorithm

S. Prabu, S. Nandakumar
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引用次数: 0

Abstract

A lot of effort has been spent over the last few years in the development of digital speech coding methods and their subsequent standardization. Algorithms have evolved which provide good quality speech at sub 8 kbps bit rates although at a much computational expense. Speech compression is proposed based on code excited linear prediction algorithm and implementation in DSP algorithm. Algorithm based on three-stage technique which involves simulate, evaluate, debug and implementation in G.723 low delay code excited linear prediction (LD-CELP)[4] algorithm. First stage, the algorithm is evaluated via simulation to determine whether it meets the design criterion. Then, it is implemented in real-time based an object oriented approach. After the algorithm is thoroughly tested, it is further refined to obtain tighter and faster coding. This technique can be applied to other real-time DSP algorithms. A simulation result shows that better speech quality is obtained. The techniques described in this paper are applicable to any other speech codec.
利用码激励线性预测算法实现实时语音压缩
在过去的几年中,人们在数字语音编码方法的发展及其随后的标准化方面付出了大量的努力。算法已经发展到可以在低于8kbps的比特率下提供高质量的语音,尽管这需要大量的计算费用。提出了基于码激励线性预测算法的语音压缩算法,并在DSP上实现。基于G.723低延迟码激励线性预测(LD-CELP)[4]算法中模拟、评估、调试和实现三阶段技术的算法。首先,通过仿真对算法进行评估,确定算法是否满足设计要求。然后,基于面向对象的方法实时实现。在对算法进行彻底测试后,进一步对其进行改进,以获得更紧凑、更快的编码。该技术可应用于其他实时DSP算法。仿真结果表明,该方法获得了较好的语音质量。本文所描述的技术适用于任何其他语音编解码器。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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