- Fix RTMPS pushes, fix RTMP push "secret" Adobe handshake
- Switch RTMP URL parsing to split at last slash rather than first slash
- "Fix" compatibility with GO-RTMP-based RTMP implementations.
- Fix compatibility with Odysee, which does a strict RTMP version check and doesn't like our version 1.2.3.4 I guess..?
- Fix HLS streamname length mismatch problems
- Prevent segfault on invalid track ID in HLS
- Fix segfault in HLS output when requesting an index for a track with exactly 1 segment available in the buffer
- Fix another HLS segfault
- Fix various incompatibilities and differences between Linux and Cygwin builds
- Make usrsctp an optional dependency
- Fix building without SSL
- Add new secure random bytes function, use it for websockets
- Switch to libsrtp2 v2.6.0 (currently latest release)
- Add patch that makes latest libsrtp2 build in latest Cygwin
- Add patch that makes srt build in latest Cygwin
- Correctly allow linking libsrtp2 and srt to local mbedtls version
Attempted to fix SRT push & embed links.
Push works, embed needs help
Changed capa to say UDP port as it is binding UDP
added methods to dtsc for output. added webrtc whip as input push url
added url_rel to DTSC
added "/" to the always_match and source_match for SDP and playlist to match with other protocols
updated Playlist to have source/always_match as array.
Noticed "variables_match" which is unused, what to do with it?
fix typo in -h controller text
- Added support for new "NowMs" field that holds up to where no new packets are guaranteed to show up, in order to lower latency.
- Added support for JSON tracks over all TS-based protocols (input and output)
- Added support for AMF metadata conversion to JSON (RTMP/FLV input)
- Fixed MP4 input subtitle tracks
- Generalized websocket-based outputs to all support the same commands and run the same core logic
- Added new "JSONLine" protocol that allows for generic direct line-by-line ingest of subtitles and/or JSON metadata tracks over a TCP socket or console standard input.
