基于DSP6437的语音采集与处理系统DSP6437-based【毕业论文】.doc

- PAGE I - 本科毕业设计 （20 届） 基于DSP6437的语音采集与处理系统DSP6437-based 所在学院 专业班级 电子信息科学与技术 学生姓名 学号 指导教师 职称 完成日期 年 月 PAGE I 摘　要 【摘要】】语音分析系统是数字信号处理最活跃的研究方向之一，在IP电话、网络视频和多媒体通信中得到了广泛的应用。一套完整的语音分析系统不仅包括语音的采集与录放，还要对复杂的语音信号进行时域和频域分析。运算量大，实时性要求高，成为信号分析领域的一大难题。DSP技术的应用为实现语音信号的采集、处理和播放奠定了基础。本文以TI公司的DEC-643为硬件平台，分析DSP处理器和Codec芯片性能特点，融合CCS3.3的软件编程功能，进行了语音采集与回放、语音编码与压缩、数字回声的生成，最后进行了基于最小均方差（LMS）算法的语音自适应滤波系统的设计。从而达到了整个语音采集与分析系统的效果。 【关键词】语音采集；TMS320DM6437；数字回声；LMS算法；自适应滤波。 Abstract 【ABSTRACT】Speech analysis system is a digital signal processing one of the most active research directions in IP telephony, video and multimedia communications networks have been widely used in. A complete voice analysis system includes not only the voice of the acquisition and playback, but also on the complex speech signal in time domain and frequency domain analysis. Large amount of computation requirements of real-time high, the field of signal analysis as a major problem. DSP technology for the realization of the speech signal acquisition, processing and playback of the foundation.In this paper, TIs DEC-6437 platform constructed to analyze the performance DSP processor and Codec chip features, integration CCS3.3 software programming, for the voice capture and playback, voice encoding and compression, digital echo Generation, and finally carried out based on the least mean square (LMS) algorithm for adaptive filtering of speech system. So as to achieve the entire speech acquisition and analysis system effects. 【KEYWORDS】Voice Acquisition; TMS320DM6437; Digital echo; LMS algorithm; Adaptive filtering. 目　录 TOC \o 1-3 \h \z \u HYPERLINK \l _Toc293644568 摘　要 PAGEREF _Toc293644568 \h II HYPERLINK \l _Toc293644569 Abstract PAGEREF _Toc293644569 \h III HYPERLINK \l _Toc293644570 目　录 PAGEREF _Toc293644570 \h IV HYPERLINK \l _Toc293644571 1 绪论 PAGEREF _Toc293644571 \h 1 HYPERLINK \l _Toc293644572 1.1