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VoIP中回声抵消算法的研究与改进

Implementation and Optimization of Echo Cancel Algorithm in VoIP

【作者】 李莉

【导师】 黄孝建;

【作者基本信息】 北京邮电大学 , 通信与信息系统, 2010, 硕士

【摘要】 随着宽带技术的迅速发展,互联网语音通信日益普及,与传统电话相比,IP电话以其网络带宽利用率高、通话成本低、提供丰富的增值功能而得到广泛应用。然而,VoIP的语音在与其他数据一起在网络中传送的时候要经过压缩、编码、打包等一系列处理,由于其基于分组交换技术,IP网络存在的数据丢包、延迟和抖动造成其QoS无法保证,其中一个急需解决的的问题就是回声抵消。在VoIP系统中不存在PSTN中由于二/四线转换器阻抗不匹配引起的线路回声,回声的主要来源是声学回声,它的产生是由于在视频会议免提情况下,听筒扬声器播放出来的声音被话筒拾取后发回远端,加上网络和数据处理等各种延迟的影响,使得远端通话者能听到自己的回声。回声的干扰会严重影响通话的质量,因此,控制和消除回声是VoIP电话的关键技术之本课题的主要工作是对普遍存在的两种回声(线路回声和声学回声)的抵消算法进行研究和改进,并在24位的AR1688 DSP芯片上实现并优化。课题的实现首先需要对算法进行定点化处理和优化,并编写汇编程序,其次进行仿真和调试。在回声抵消算法通过调试后,为了降低算法的时间和空间复杂度,提高运行速度,适应低速处理器的要求,降低设备功耗,利用了AR1688芯片自身的硬件特点和指令特点,对算法进行优化,最终在以AR1688为核心的IP话机上成功地实现了回声抵消的要求。

【Abstract】 With the advent of IP voice technology, VoIP (Voice over IP) receives more and more application in recent years. Compared with the traditional telephone, IP phones are widely used because its high network bandwidth utilization, low-cost and flexibility to provide value-added features. However, the quality of voice can not be guaranteed because Internet is the best-effort model. If IP phone wants to compete with traditional telephone, it must improve its voice quality. One of key technologies is echo cancellation in video conference.There are two types of echo in telephone network, the first type, line echo can be generated because of the mismatch between the subscriber loop and switch office where two wires and four wires are connected. The second type, acoustic echo arises with the invention of hand-free phones, in which the microphone and loudspeaker are coupled. Acoustic echo is characterized with long delay and complex and unstable environment, and need more genius algorithms.The core of echo cancellation is adaptive filter theory, according to which residual echo is got by subtracting the echo replica from the actual echo and is further used to adapt the coefficients of the filter dynamically. Traditional adaptive filter algorithm has achieved good performance in dealing with line echo which has short delay. To handle acoustic echo, large-taps filters in frequency domain are often used, which make good use of FFT’s efficiency to meet the real time requirements.This thesis is to investigate the line echo cancellation and acoustic echo cancellation algorithm, multidelay block frequency domain adaptive filter (MDF) and expects to implement and optimization their codec on AR1688, a fixed-point digital signal processor. It is meaningful to realize the AEC codec in AR1688 because of the low cost and high precision. The implementation was in two steps:first, make the float point to fixed-point conversion in C with optimization. Then translate the C codecs to DSP assembly codes, which is compatible with ADSP2181, debug the echo cancellation codec, the next work is the optimization of the echo cancellation codec according to the hardware features and instruction characteristic of AR1688 chip in order to reduce the complexity of algorithm, improve the speed of program, adapt the requirements of processor, and reduce the power of equipment. Eventually, we successfully achieved the requirements of real-time AEC on IP phone which the core element is AR1688 chip.

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