* Add optional idle heartbeat for Birdseye (periodic frame emission when idle)
birdseye: add optional idle heartbeat and FFmpeg tuning envs (default off)
This adds an optional configuration field `birdseye.idle_heartbeat_fps` to
enable a lightweight idle heartbeat mechanism in Birdseye. When set to a value
greater than 0, Birdseye periodically re-sends the last composed frame during
idle periods (no motion or active updates).
This helps downstream consumers such as go2rtc, Alexa, or Scrypted to attach
faster and maintain a low-latency RTSP stream when the system is idle.
Key details:
- Config-based (`birdseye.idle_heartbeat_fps`), default `0` (disabled).
- Uses existing Birdseye rendering pipeline; minimal performance impact.
- Does not alter behavior when unset.
Documentation: added tip section in docs/configuration/restream.md.
* Update docs/docs/configuration/restream.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Update docs/docs/configuration/reference.md
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
* Refactors Birdseye idle frame broadcasting
Simplifies the idle frame broadcasting logic by removing the dedicated thread.
The idle frame is now resent directly within the main loop,
improving efficiency and reducing complexity. Also, limits the idle
heartbeat FPS to a maximum of 10 since the framebuffer is limited to 10 anyway
* ruff fix
---------
Co-authored-by: Nicolas Mowen <nickmowen213@gmail.com>
Co-authored-by: Francesco Durighetto <francesco.durighetto@subbyx.com>
Co-authored-by: duri <duri@homelabubuntu.durihome.unifi>
* 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>
