纯软件视频会议编解码器在PC机上的实现

A. Dvorkovich, V. P. Dvorkovich, G.N. Mokhin, A.Y. Sokolov
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引用次数: 0

摘要

近年来,计算机技术的飞速发展为以数字形式传输音视频信息提供了新的解决方案。通过交换公共电话线、移动无线电系统、本地计算机网络、互联网和卫星电信等容量有限的渠道传输数字音视频信号,引起了人们日益增长的兴趣。这种线路中可用于传输音频和视频信息的比特率受限于特定应用,范围从30到512 Kbit/s。尽管低比特率数字编码本身需要大量的计算资源,并且对实时编码器的性能有严重的限制。还应该考虑到,对于具有低比特率的信道,编解码器必须实现压缩算法,使我们能够实现大的压缩比和可接受的质量。需要的压缩比只能通过帧内编码和帧间运动补偿来实现。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
On the implementation of software-only videoconferencing codec on PC
Intensive recent progress in computer technologies has stimulated new solutions for transmission of audio and video information in digital form, The transmission of digital video and audio signals through channels of restricted capacity, such as switched public telephone lines, mobile radio systems, local computer networks, Internet and satellite telecommunications attract a constantly growing interest. The bit rate in such lines available for transmission of audio and video information is limited dependent upon particular application in the range from 30 to 512 Kbit/s. In spite of the low bit rate digital coding itself requires significant computational resources and implies severe limitations on the performance of the real-time encoder. It should be also taken into account that for channels with low bit rate a codec must implement compression algorithms allowing us to achieve a large compression ratio and acceptable quality. Necessary compression ratios can be achieved only by means of intraframe coding as well as interframe motion compensation.
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