AMBE语音压缩算法实现及改进  

The Implement and Improvement of AMBE Voice Compressing Algorithm

作　　者：陆奕奕[1] 袁三男[1] 宋文涛[1] 

机构地区：[1]上海交通大学电子工程系,200030

出　　处：《通信技术》1998年第4期63-66,共4页Communications Technology

摘　　要：文章研究了AMBE语音压缩算法（AdvancedMulti－bandExcitationSpeechCoder）及其在数字信号处理器（DSP）上的实现。为了适应实际应用的需要，对AMBE算法进行了改进，去除了其FEC部分，并在DSP上实现了无FEC的AMBE处理程序，使数据传输速率从4．8kbps降至3．6kbps，从而使语音压缩率更高。实验结果证明话音质量基本不受影响，而所需信道速率降低了25％。This paper studies the AMBE algorithm and its implement on the Digital SignalProcessor(DSP). In order to increase the channel utilization and meet the need of practical application, the AMBE algorithm is improved by taking out FEC module. Furthermore, the processing programs on DSP are also carried out without FEC. Thus, the speed of data transmission decreases from 4. 8kbps to 3 .6kbps and the voice compression ratio become even higher. Experiments show that the voice quality is hardly influenced, while the channel speed needed is decreased by 25%.[

关 键 词：语音压缩 编码 解码 AMBE 

分 类 号：TN912.3[电子电信—通信与信息系统]