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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>
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@@ -5,7 +5,7 @@ import pytest
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from facefusion import process_manager
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from facefusion.download import conditional_download
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from facefusion.ffprobe import detect_audio_channel_total, detect_audio_frame_total, detect_audio_sample_rate
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from .helper import get_test_example_file, get_test_examples_directory
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from .assert_helper import get_test_example_file, get_test_examples_directory
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@pytest.fixture(scope = 'module', autouse = True)
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