* reload the window on 401
* backend apis for auth
* add login page
* re-enable web linter
* fix login page routing
* bypass csrf for internal auth endpoint
* disable healthcheck in devcontainer target
* include login page in vite build
* redirect to login page on 401
* implement config for users and settings
* implement JWT actual secret
* add brute force protection on login
* add support for redirecting from auth failures on api calls
* return location for redirect
* default cookie name should pass regex test
* set hash iterations to current OWASP recommendation
* move users to database instead of config
* config option to reset admin password on startup
* user management UI
* check for deleted user on refresh
* validate username and fixes
* remove password constraint
* cleanup
* fix user check on refresh
* web fixes
* implement auth via new external port
* use x-forwarded-for to rate limit login attempts by ip
* implement logout and profile
* fixes
* lint fixes
* add support for user passthru from upstream proxies
* add support for specifying a logout url
* add documentation
* Update docs/docs/configuration/authentication.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Update docs/docs/configuration/authentication.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
---------
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Initial audio classification model implementation
* fix mypy
* Keep audio labelmap local
* Cleanup
* Start adding config for audio
* Add the detector
* Add audio detection process keypoints
* Build out base config
* Load labelmap correctly
* Fix config bugs
* Start audio process
* Fix startup issues
* Try to cleanup restarting
* Add ffmpeg input args
* Get audio detection working
* Save event to db
* End events if not heard for 30 seconds
* Use not heard config
* Stop ffmpeg when shutting down
* Fixes
* End events correctly
* Use api instead of event queue to save audio events
* Get events working
* Close threads when stop event is sent
* remove unused
* Only start audio process if at least one camera is enabled
* Add const for float
* Cleanup labelmap
* Add audio icon in frontend
* Add ability to toggle audio with mqtt
* Set initial audio value
* Fix audio enabling
* Close logpipe
* Isort
* Formatting
* Fix web tests
* Fix web tests
* Handle cases where args are a string
* Remove log
* Cleanup process close
* Use correct field
* Simplify if statement
* Use var for localhost
* Add audio detectors docs
* Add restream docs to mention audio detection
* Add full config docs
* Fix links to other docs
---------
Co-authored-by: Jason Hunter <hunterjm@gmail.com>
* Update go2rtc to 1.3.0
* Increment to 1.3.1
* Increment to 1.3.2
* Update webrtc player to match latest
* Update version to 1.4.0
* Update mse player
* Update birdseye mse player
* remove logs
* Update docs to link to new version
* Final web lint fixes
* Update versions
* Update live.md
Placed `ffmpeg:http_cam#audio=opus` in quotes so it doesn't appear as commented out in docs.
* Update restream.md
Placed `ffmpeg:http_cam#audio=opus` in quotes so it doesn't appear as commented out in docs.
* Add video codec to restream config
* Add handling of encode engine and video codec
* Add test for video encoding
* Set in main configuration docs as well
* Add example to restream docs
* Put back patch
* Try using RTSP for restream
* Add ability to get snapshot of birdseye when birdseye restream is enabled
* Write to pipe instead of encoding mpeg1
* Write to cache instead
* Use const for location
* Formatting
* Add hardware encoding for birdseye based on ffmpeg preset
* Provide framerate
* Adjust args
* Fix order
* Delete pipe file if it exists
* Cleanup spacing
* Fix spacing
* Start restream before detection
* Add docs explaining how to reduce connections to the camera
* Fix typos for consistency
* Add link to other part of doc for readability
* 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>
