* fix stdin close error * Refactor stream endpoint, fix encoder thread safety and improve tests * fix and improve test * remove not None * use Enum * use Enum and add todo * remove poll
* implement basic webrtc_stream * add aiortc to requirements.txt * update aiortc version * rename variables with rtc_ prefix * changes * changes * change helper to assert_helper and stream_helper * rename variables with rtc_ prefix * add error handling * return whole connection * remove monkey patch and some cleaning * cleanup * tiny adjustments * tiny adjustments * proper typing and naming for rtc offer set * - remove async from on_video_track method - rename source -> target - add audio * audio always before video --------- Co-authored-by: henryruhs <info@henryruhs.com>
henryruhs