- Add server stats reset and frame drop recovery functions
- Implement global audio server instance management
- Add WebRTC audio track replacement capability
- Improve audio relay initialization with retry logic
- Enhance quality change handling with adaptive buffer management
- Add global helper functions for audio quality control
- Replace MuteMicrophone calls with StartMicrophone/StopMicrophone for clearer behavior
- Update microphone state broadcasting to reflect actual subprocess status
- Modify UI to use enable/disable terminology instead of mute/unmute
- Ensure microphone device changes properly restart the active microphone
- Implement new POST /microphone/stop endpoint
- Refactor mute handling to properly start/stop audio processes
- Add callback mechanism for audio relay to reconnect to current session
- Simplify UI microphone controls by combining mute/start-stop functionality
- Extract audio-related handlers into separate file for better organization
- Simplify session creation logic by removing duplicate code paths
- Add new Prometheus metrics for connection monitoring
- Reduce websocket ping interval from 30s to 15s for better responsiveness
Alex P