* fix typo in RtspSession.cpp

* Trying to send mjpeg via MultiMediaSourceMuxer

* Improved JPEGRtpEncoder::inputFrame code but still not working

* 优化代码

* 完善jpeg相关逻辑

* Micro fix

* FrameJPEG renamed to JPEGFrame according to ZLM style

* Modified  JPEGRtpEncoder::inputFrame and JPEGRtpEncoder::rtp_send_jpeg

* getVideoHeight(), getVideoWidth() and getVideoFps() in JPEGTrack

* mjpeg rtp打包避免内存拷贝/修复mjpeg rtp解包huffman_table size字段错误的bug

* 支持mjpeg pix type

* 优化性能

* add bom header

* 优化MultiMediaSourceMuxer头文件包含

* 将MediaSinkDelegate和Demux移到MediaSink中

* MediaSource头文件重构, 独立出PacketCache.h
精简Frame和Track的头文件

* Rtmp头文件重构

* Rtsp头文件重构

* webrtc头文件重构

* 规范.h头文件包含，并将其移到.cpp中:
- 尽量不包含Common\config.h
- Util\File.h
- Rtsp/RtspPlayer.h
- Rtmp/RtmpPlayer.h

* 删除多余的Stamp.h和Base64包含

bye(203) 会在 sender audio track mute 时出现, 不能作为 shutdown 的依据

* webrtc push/play支持通过http参数指定tcp

* force_tcp改成perferred_tcp

Co-authored-by: xiongziliang <771730766@qq.com>

* webrtc server/session/cadidate 改为tcp

* 先屏蔽检查isCurrentThread

* 接受和发送的数据处理tcp 2字节头

* 处理rtc tcp 分片

* 完善webrtc over tcp

* 精简rtp服务器相关代码

* 适配webrtc AV1编码: #2091

* webrtc tcp模式支持Firefox

* webrtc tcp模式支持线程安全

* c sdk支持webrtc tcp

Co-authored-by: ziyue <1213642868@qq.com>

整理代码

* fix c2065 and c2051 error on windows, and the head file mpeg-ts-proto.h is also deprecated

* update media-server to 357ef88