Implement basic webrtc stream (#1054)

* 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>

tests/stream_helper.py

@@ -0,0 +1,21 @@
+from aiortc import RTCPeerConnection, VideoStreamTrack
+
+from facefusion.types import RtcOfferSet
+
+
+async def create_rtc_offer() -> RtcOfferSet:
+	rtc_connection = RTCPeerConnection()
+	rtc_connection.addTrack(VideoStreamTrack())
+	rtc_offer = await rtc_connection.createOffer()
+
+	await rtc_connection.setLocalDescription(rtc_offer)
+
+	rtc_offer_set : RtcOfferSet =\
+	{
+		'sdp': rtc_connection.localDescription.sdp,
+		'type': rtc_connection.localDescription.type
+	}
+
+	await rtc_connection.close()
+
+	return rtc_offer_set