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lalext's Introduction

lalext

🇨🇳

lal_website  ...存放lal官方文档页面的文档 https://pengrl.com/lal
thirdparty   ...第三方c/c++库(目前主要是ffmpeg及其依赖),包含编译脚本
rfc          ...标准协议文档

avfile       ...存放测试用的音视频文件
rtmp2webrtc  ...rtmp转webrtc的网关,上游可以接入任意的rtmp协议服务器。目前基本不维护了,建议去关注lalmax(https://github.com/q191201771/lalmax)
srt          ...基于Lalserver支持srt协议,目前已移入lalmax(https://github.com/q191201771/lalmax)管理

更具体的说明,见子目录下的文档。

🇬🇧

List of subdirectories on this repository

Note: translations to English are still in early stages; in case of doubt, the Chinese version shall always prevail.

  • lal_website — Files for the official lal documentation page at https://pengrl.com/lal.
  • lal_website_english — Ongoing effort of translating the documentation files into English.
  • thirdparty — Third-party C/C++ libraries (currently mainly ffmpeg and its dependencies), including compilation scripts.
  • rfc — Documentation for the many protocol standards.
  • avfile — Audio and video files for testing (free).
  • rtmp2webrtcrtmp to WebRTC gateway, upstream can access any rtmp protocol server. Currently basically not maintained, it is recommended to follow lalmax.
  • srt — Based on Lalserver support srt protocol, currently moved to lalmax management.

For more specific instructions, see the documentation in the subdirectory.

lalext's People

Contributors

bean-du avatar captainneil avatar gwynethllewelyn avatar lio19 avatar q191201771 avatar thewind296 avatar zsc714725 avatar

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lalext's Issues

lalext ice修改为lite ice

因为lalext的场景都是作为服务端,必然会有公网ip。在这种情况下我们可以用lite ice
可以优化:

  1. 不用访问stun服务的去获取公网服务器的ip。减少对stun服务的依赖。
  2. 减少candidate获取时间,将获取candidate的时间减少为0,大大降低协商时间。

支持rtsp2webrtc,支持aac音频转opus

  • [feat] 支持基础功能 #9
  • [opt] 使用rtp直接转rtp,而不是rtp合帧后再切割成rtp
  • [doc] 更新相关的文档
  • [refactor] rtsp集成在rtmp2webrtc中,相关的命名需要重构
  • [opt] aac音频转opus依赖cgo,需要支持指定是否开启(比如条件编译),从而让不想用cgo的业务方可以继续使用没声音的版本

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