mirror of
https://github.com/dragonpilot/dragonpilot.git
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f058b5d64e
* WebRTCClient and WebRTCServer abstractions * webrtc client implementation * Interactive test scripts * Send localDescriptions as offer/asnwer, as they are different * Tracks need to be added after setting remote description for multi-cam streaming to work * Remove WebRTCStreamingMetadata * Wait for tracks * Move stuff to separate files, rename some things * Refactor everything, create WebRTCStreamBuilder for both offer and answers * ta flight done time to grind * wait for incoming tracks and channels * Dummy track and frame reader track. Fix timing. * dt based on camera type * first trial of the new api * Fix audio track * methods for checking for incoming tracks * Web migration part 2 * Fixes for stream api * use rtc description for web.py * experimental cereal proxy * remove old code from bodyav * fix is_started * serialize session description * fix audio * messaging channel wrapper * fix audiotrack * h264 codec preference * Add codec preference to tracks * override sdp codecs * add logging * Move cli stuff to separate file * slight cleanup * Fix audio track * create codec_mime inside force_codec function * fix incoming media estimation * move builders to __init__ * stream updates following builders * Update example script * web.py support for new builder * web speaker fixes * StreamingMediaInfo API * Move things around * should_add_data_channel rename * is_connected_and_ready * fix linter errors * make cli executable * remove dumb comments * logging support * fix parse_info_from_offer * improve type annotations * satisfy linters * Support for waiting for disconnection * Split device tracks into video/audio files. Move audio speaker to audio.py * default dt for dummy video track * Fix cli * new speaker fixes * Remove almost all functionality from web.py * webrtcd * continue refactoring web.py * after handling joystick reset in controlsd with #30409, controls are not necessary anymore * ping endpoint * Update js files to at least support what worked previously * Fixes after some tests on the body * Streaming fixes * Remove the use of WebRTCStreamBuilder. Subclass use is now required * Add todo * delete all streams on shutdown * Replace lastPing with lastChannelMessageTime * Update ping text only if rtc is still on * That should affect the chart too * Fix paths in web * use protocol in SSLContext * remove warnings since aiortc is not used directly anymore * check if task is done in stop * remove channel handler wrapper, since theres only one channel * Move things around * Moved webrtc abstractions to separate repository * Moved webrtcd to tools/webrtc * Update imports * Add bodyrtc as dependency * Add webrtcd to process_config * Remove usage of DummyVideoStreamTrack * Add main to webrtcd * Move webrtcd to system * Fix imports * Move cereal proxy logic outside of runner * Incoming proxy abstractions * Add some tests * Make it executable * Fix process config * Fix imports * Additional tests. Add tests to pyproject.toml * Update poetry lock * New line * Bump aiortc to 1.6.0 * Added teleoprtc_repo as submodule, and linked its source dir * Add init file to webrtc module * Handle aiortc warnings * Ignore deprecation warnings * Ignore resource warning too * Ignore the warnings * find free port for test_webrtcd * Start process inside the test case * random sleep test * test 2 * Test endpoint function instead * Update comment * Add system/webrtc to release * default arguments for body fields * Add teleoprtc to release * Bump teleoprtc * Exclude teleoprtc from static analysis * Use separate event loop for stream session tests
61 lines
2.2 KiB
Python
Executable File
61 lines
2.2 KiB
Python
Executable File
#!/usr/bin/env python
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import asyncio
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import json
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import unittest
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from unittest.mock import MagicMock, AsyncMock
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# for aiortc and its dependencies
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import warnings
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warnings.filterwarnings("ignore", category=DeprecationWarning)
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from openpilot.system.webrtc.webrtcd import get_stream
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import aiortc
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from teleoprtc import WebRTCOfferBuilder
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class TestWebrtcdProc(unittest.IsolatedAsyncioTestCase):
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async def assertCompletesWithTimeout(self, awaitable, timeout=1):
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try:
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async with asyncio.timeout(timeout):
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await awaitable
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except asyncio.TimeoutError:
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self.fail("Timeout while waiting for awaitable to complete")
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async def test_webrtcd(self):
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mock_request = MagicMock()
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async def connect(offer):
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body = {'sdp': offer.sdp, 'cameras': offer.video, 'bridge_services_in': [], 'bridge_services_out': []}
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mock_request.json.side_effect = AsyncMock(return_value=body)
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response = await get_stream(mock_request)
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response_json = json.loads(response.text)
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return aiortc.RTCSessionDescription(**response_json)
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builder = WebRTCOfferBuilder(connect)
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builder.offer_to_receive_video_stream("road")
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builder.offer_to_receive_audio_stream()
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builder.add_messaging()
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stream = builder.stream()
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await self.assertCompletesWithTimeout(stream.start())
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await self.assertCompletesWithTimeout(stream.wait_for_connection())
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self.assertTrue(stream.has_incoming_video_track("road"))
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self.assertTrue(stream.has_incoming_audio_track())
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self.assertTrue(stream.has_messaging_channel())
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video_track, audio_track = stream.get_incoming_video_track("road"), stream.get_incoming_audio_track()
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await self.assertCompletesWithTimeout(video_track.recv())
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await self.assertCompletesWithTimeout(audio_track.recv())
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await self.assertCompletesWithTimeout(stream.stop())
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# cleanup, very implementation specific, test may break if it changes
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self.assertTrue(mock_request.app["streams"].__setitem__.called, "Implementation changed, please update this test")
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_, session = mock_request.app["streams"].__setitem__.call_args.args
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await self.assertCompletesWithTimeout(session.post_run_cleanup())
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if __name__ == "__main__":
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unittest.main()
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