G.729语音压缩编码算法研究与DSP实现

发布时间：2018-10-15 17:22

【摘要】：G.729是国际电信联盟提出的语音编码标准。它具有较低的编码速率、较低的延时和较高的合成语音质量,因此在语音通信中得到了广泛的应用。目前,G.729语音编码算法已经很成熟,但仍然存在如下问题:首先,G.729语音编码算法复杂度较高,对语音通信来说,实时性是其中的一个关键因素,现有标准算法很难直接被使用。其次,虽然G.729标准已提出了多年,但是集成的G.729编解码器出现的相对较少。为此,论文以G.729标准为基础,以项目组自主研发的手持式信息平台为背景,对G.729标准进行研究和实现。第一,研究G.729语音编码算法。通过对G.729标准算法进行分析和仿真,得到编解码器的算法复杂度,为能够进行实时编解码提供依据。第二,设计语音编解码算法DSP软件实现架构。论文选用DM3730双核芯片的DSP核与音频芯片,设计语音采集与播放模块,语音编解码模块和中断服务程序模块,并对程序从编译器级和代码级两方面进行优化,保证G.729编解码算法能够实时处理语音信号。第三,验证G.729编解码算法的性能。以手持式信息平台为硬件背景,通过DSP在线调试工具对音频采集与播放模块和语音编解码算法模块进行测试。合成语音质量的平均分段信噪比为10dB,编码一帧语音信号需要5.4ms,解码一帧语音信号需要1.7ms,满足实时性的要求,达到了预期的目标。本文研究的内容对在DSP平台上实现语音信号采集与编解码的设计与实现提供了解决方案,具有一定的实用性。
[Abstract]:G. 729 is a speech coding standard proposed by the International Telecommunication Union. Because of its low coding rate, low delay and high synthetic speech quality, it has been widely used in speech communication. At present, G. 729 speech coding algorithm has been very mature, but still has the following problems: first, G. 729 speech coding algorithm complexity is high, for speech communication, real-time is one of the key factors, the existing standard algorithm is very difficult to be directly used. Secondly, although the G.729 standard has been proposed for many years, there are relatively few integrated G.729 codecs. Therefore, based on the G.729 standard and the handheld information platform developed by the project team, the thesis researches and implements the G.729 standard. Firstly, G. 729 speech coding algorithm is studied. Based on the analysis and simulation of G. 729 standard algorithm, the algorithm complexity of the codec is obtained, which provides the basis for real-time coding and decoding. Second, the design of speech codec algorithm DSP software architecture. In this paper, the DSP core and audio chip of DM3730 dual-core chip are selected to design voice acquisition and playback module, speech codec module and interrupt service program module. The program is optimized from compiler level and code level. The G. 729 codec algorithm can process the speech signal in real time. Thirdly, the performance of G. 729 codec algorithm is verified. Taking the handheld information platform as the hardware background, the audio acquisition and playback module and the speech codec algorithm module are tested by the DSP online debugging tool. The average segmented signal-to-noise ratio (SNR) of the synthesized speech is 10 dB, the coding of a frame of speech signal needs 5.4 Ms, and the decoding of a frame of speech signal needs 1.7 Ms, which meets the real-time requirements and achieves the expected goal. The content of this paper provides a solution for the design and implementation of speech signal acquisition and codec on DSP platform.
【学位授予单位】：电子科技大学
【学位级别】：硕士
【学位授予年份】：2014
【分类号】：TN912.3

【共引文献】