* Implement renaming in model editing dialog
* add transcription faq
* remove incorrect constraint for viewer as username
should be able to change anyone's role other than admin
* Don't save redundant state changes
* prevent crash when a camera doesn't support onvif imaging service required for focus support
* Fine tune behavior
* Stop redundant go2rtc stream metadata requests and defer audio information to allow bandwidth for image requests
* Improve cleanup logic for capture process
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Co-authored-by: Josh Hawkins <32435876+hawkeye217@users.noreply.github.com>
* Don't add to history when opening search dialog
* Update caniuse
* Revamp the history handling for dialog components
* clarify audio transcription docs
* Use titlecase helper
* Allow running object clasasification on stationary objects
* small spacing tweaks for tablets
* require admin role to delete users
* explicitly prevent deletion of admin user
---------
Co-authored-by: Josh Hawkins <32435876+hawkeye217@users.noreply.github.com>
* fix wording in reference config
* spacing tweaks
* make live view settings drawer scrollable
* clarify audio transcription docs
* change audio transcription icon to activity indicator when transcription is in progress
the backend doesn't implement any kind of queueing for speech event transcription
* tracking details tweaks
- Add attribute box overlay and area
- Add score
- Throttle swr revalidation during video component rerendering
* add mse codecs to console debug on errors
* add camera name
* camera level config
* set up model runner on thread start to avoid unpickling error
* ensure feature is enabled globally
* suppress info logs from faster_whisper
* fix incorrect event_type for api and audio timeline entries
* docs
* fix
* clean up
* install new packages for transcription support
* add config options
* audio maintainer modifications to support transcription
* pass main config to audio process
* embeddings support
* api and transcription post processor
* embeddings maintainer support for post processor
* live audio transcription with sherpa and faster-whisper
* update dispatcher with live transcription topic
* frontend websocket
* frontend live transcription
* frontend changes for speech events
* i18n changes
* docs
* mqtt docs
* fix linter
* use float16 and small model on gpu for real-time
* fix return value and use requestor to embed description instead of passing embeddings
* run real-time transcription in its own thread
* tweaks
* publish live transcriptions on their own topic instead of tracked_object_update
* config validator and docs
* clarify docs
* 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>
