* randomize internal seg list segments
* random
* pytest-randomly sets random.seed to a consistent value for all the workers/processes
* noeol
* update
* Revert "update"
This reverts commit aff9a69c4e.
* lock
* don't randomize by default
* remove random-order
* strict
* random
* one fix
* test
* does nothing
* rm tests
* Revert "rm tests"
This reverts commit b548e3fcd4.
* (can't repro locally) just athena should be fine
* bs1
* bs2
* bs3
* bs4
* bs5
* wrong way
* no controls
* no car
* no board
* controls?
* crazy -common
* Revert "crazy -common"
This reverts commit 02365d712b.
* test athena
* test athena 2
* test athena 3
* test athena 4
* test athena 5
* test athena 6
* test athena 7
* test athena 8
* test athena 9
* ??
* in one commit
* common?
* car and board
* -controls -board
* random-order
* no board
* revert
* car/tests
* least likely
* try
* try 2
* draft
* draft
* so much better
* cmt
* use randomly
* not needed here
* directly modify option works
* bb
* test time
* Revert "test time"
This reverts commit 2c5caabe2b.
* tmut
* i concur
* revert
* 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
commaci-public
Maxime Desroches
Adeeb Shihadeh
Hoang Bui
Mauricio Alvarez Leon
cl0cks4fe
Shane Smiskol
YassineYousfi
Justin Newberry
royjr
Kacper Rączy