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Refactor/rtc cleanup 3 (#1118)

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* tweak rtc store and make the decision to ban trivial testing

* clear todos for rtc_test, remove redundant tests

* clear todos for rtc_test, remove redundant tests

* break negotiation out of rtc flow, introduce create_sdp_answer and set_remote_description

* add todo

* move timeline control to the stream helper, clean send_audio|video_to_peers

* rename some methods

* fix test

* introduce detect_sdp_media

* introduce detect_sdp_media
This commit is contained in:
Henry Ruhs
2026-05-16 09:06:04 +02:00
committed by GitHub
parent 95435f842c
commit dd1ded1408
10 changed files with 157 additions and 219 deletions
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tests/test_stream_helper.py
+3 -3
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@@ -209,8 +209,8 @@ def test_handle_video_stream() -> None:
websocket.accept.assert_called_once_with(subprotocol = 'proto')
websocket.send_text.assert_called_once_with('ready')
websocket.close.assert_called_once()
mock_rtc.create_rtc_peers.assert_called_once_with('session-1')
mock_rtc.destroy_rtc_peers.assert_called_once_with('session-1')
mock_rtc.init_peers.assert_called_once_with('session-1')
mock_rtc.delete_peers.assert_called_once_with('session-1')
_, loop_session_id, loop_resolution = mock_loop.call_args[0]
assert loop_session_id == 'session-1'
assert loop_resolution == (64, 64)
@@ -224,7 +224,7 @@ def test_handle_video_stream() -> None:
asyncio.run(handle_video_stream(websocket))
websocket.accept.assert_called_once()
websocket.send_text.assert_not_called()
mock_rtc.create_rtc_peers.assert_not_called()
mock_rtc.init_peers.assert_not_called()
websocket = _make_handler_websocket([ {'type': 'websocket.receive', 'bytes': video_packet}, {'type': 'websocket.receive', 'bytes': audio_packet}, {'type': 'websocket.disconnect'} ])
with patch('facefusion.apis.stream_helper.get_sec_websocket_protocol', return_value = 'proto'), \