* Check ffmpeg version instead of checking for presence of BTBN_PATH
* Query ffmpeg version in s6 run script instead of subprocessing in every import
* Define LIBAVFORMAT_VERSION_MAJOR in devcontainer too
* Formatting
* Default ffmpeg version to current btbn version so unit tests pass
* Add isort and ruff linter
Both linters are pretty common among modern python code bases.
The isort tool provides stable sorting and grouping, as well as pruning
of unused imports.
Ruff is a modern linter, that is very fast due to being written in rust.
It can detect many common issues in a python codebase.
Removes the pylint dev requirement, since ruff replaces it.
* treewide: fix issues detected by ruff
* treewide: fix bare except clauses
* .devcontainer: Set up isort
* treewide: optimize imports
* treewide: apply black
* treewide: make regex patterns raw strings
This is necessary for escape sequences to be properly recognized.
* Upgrade s6-overlay from 3.1.3.0 to 3.1.4.0
* Add go2rtc healthcheck service
Also don't make go2rtc exits cause the container to fail.
* Reword healthcheck message
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Add timeout to go2rtc healthcheck
* Update healthcheck message
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Give additional time for go2rtc start/restart
* Fix typo
* Avoid creating go2rtc config multiple times
* Fix healthcheck not starting
* Fix sleep
* Fix more hidden logs
* Decrease time window and use curl's timeout flag
---------
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Auto discover internal WebRTC candidate for add-on
* Write logs to stderr
* Fix port number
* Integrate with newest changes
* Update docs
* Use local variable more
* Use Python to write file, fix JSON->YAML
* Store into variable
* Update docs/docs/configuration/live.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Update docs/docs/configuration/live.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Update docs/docs/configuration/live.md
* Update docs/docs/configuration/live.md
* Refator s6 scripts to the new format
* Remove unneeded workaround
* Update docker/rootfs/usr/local/go2rtc/create_config.py
* Migrate logging to new s6 format
* Remove more unnecessary s6 variables
* Fix prepare-log and when go2rtc is not present in config
* Restart the whole container if either Frigate or go2rtc fails
* D
* Fix service name in finish
* Fix nginx finish comment
* Restart improvements
* Fix devcontainer
* Fix format
* Update Dockerfile
Co-authored-by: Felipe Santos <felipecassiors@gmail.com>
* Improve scripts logging
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Pull go2rtc dependency
* Add go2rtc to local services and add to s6
* Add relay controller for go2rtc
* Add restream role
* Add restream role
* Add restream to nginx
* Add camera live source config
* Disable RTMP by default and use restream
* Use go2rtc for camera config
* Fix go2rtc move
* Start restream on frigate start
* Send restream to camera level
* Fix restream
* Make sure jsmpeg works as expected
* Make view rspect live size config
* Tweak player options to fit live view
* Adjust VideoPlayer to accept live option which disables irrelevant controls
* Add multiple options from restream live view
* Add base for webrtc option
* Setup specific restream modules
* Make mp4 the default streaming for now
* Expose 8554 for rtsp relay from go2rtc
* Formatting
* Update docs to suggest new restream method.
* Update docs to reflect restream role
* Update docs to reflect restream role
* Add webrtc player
* Improvements to webRTC
* Support webrtc
* Cleanup
* Adjust rtmp test and add restream test
* Fix tests
* Add restream tests
* Add live view docs and show different options
* Small docs tweak
* Support all stream types
* Update to beta 9 of go2rtc
* Formatting
* Make jsmpeg the default
* Support wss if made from https
* Support wss if made from https
* Use onEffect
* Set url outside onEffect
* Fix passed deps
* Update docs about required host mode
* Try memo instead
* Close websocket on changing camera
* Formatting
* Close pc connection
* Set video source to null on cleanup
* Use full path since go2rtc can't see PATH var
* Adjust audio codec to enable browser audio by default
* Cleanup stream creation
* Add restream tests
* Format tests
* Mock requests
* Adjust paths
* Move stream configs to restream
* Remove live source
* Remove live config
* Use live persistence for which view to use on each camera
* Fix live sizes
* Only use jsmpeg sizes for jsmpeg live
* Set max live size
* Remove access of live config
* Add selector for live view source in web view
* Remove RTMP from default list of roles
* Update docs
* Fix tests
* Fix docs for live view modes
* make default undefined to avoid race condition
* Wait until camera source is loaded to avoid race condition
* Fix tests
* Add config to go2rtc
* Work with config
* Set full path for config
* Set to use stun
* Check for mounted file
* Look for frigate-go2rtc
* Update docs to reflect webRTC configuration.
* Add link to go2rtc config
* Update docs to be more clear
* Update docs to be more clear
* Update format
Co-authored-by: Felipe Santos <felipecassiors@gmail.com>
* Update live docs
* Improve bash startup script
* Add option to force audio compatibility
* Formatting
* Fix mapping
* Fix broken link
* Update go2rtc version
* Get go2rtc webui working
* Add support for mse
* Remove mp4 option
* Undo changes to video player
* Update docs for new live view options
* Make separate path for mse
* Remove unused
* Remove mp4 path
* Try to get go2rtc proxy working
* Try to get go2rtc proxy working
* Remove unused callback
* Allow websocket on restrea dashboard
* Make mse default stream option
* Fix mse sizing
* don't assume roles is defined
* Remove nginx mapping to go2rtc ui
Co-authored-by: Felipe Santos <felipecassiors@gmail.com>
Co-authored-by: Blake Blackshear <blakeb@blakeshome.com>
