feat: Complete fresh restart with clean Streamlit configuration

debug: Add print statement to test Python execution before Streamlit

fix: Ultra-minimal hello_world.py to diagnose 500 errors

fix: Remove .streamlit/config.toml and simplify to minimal hello_world test

Merge remote-tracking branch 'hf/main'

restore: Bring back full VoiceCal WebRTC interface

fix: Add explicit Streamlit configuration for HF Spaces health check

debug: Add hello world test for HF Spaces restart cycle investigation

debug: Switch to ultra-minimal Streamlit test to isolate restart loop issue

debug: Simplify VoiceCal to isolate restart issue cause

fix: Resolve VoiceCal restart loop by removing conflicting Dockerfile

refactor: Streamline VoiceCal to pure WebSocket connectivity with STT v1.0.0

fix: Switch to Gradio HTTP API fallback for WebSocket connectivity issues

fix: Update WebSocket URL to /api/ws/stt to avoid Gradio routing conflicts

fix: Force Streamlit to use port 7860 for HuggingFace Spaces

fix: Shorten description to meet HF Spaces 60 character limit

fix: Simplify VoiceCal to single Streamlit service with direct STT connection

feat: Implement WebRTC integration following unmute.sh pattern

fix: Remove .gitattributes file that triggers HF VS Code detection

fix: Switch to Streamlit SDK to bypass VS Code detection

fix: Simplify VoiceCal build to avoid VS Code detection

fix: Add build dependencies for webrtcvad and audio libraries

fix: Switch to Docker SDK to avoid VS Code extension build errors

fix: Resolve syntax error in WebSocket retry logic

feat: Add comprehensive WebSocket retry logic with GPU cold start timing calibration

feat: Add STT service preload to prevent cold start delays

feat: Update to v0.8.2 with WebSocket STT integration

fix: Resolve Gradio client FileData validation error

fix: Improve Gradio client STT endpoint discovery

fix: Always initialize WebSocket URLs regardless of MCP status

debug: Add WebSocket URL availability debugging to STT service

feat: Update version to v0.8.0 - Whisper corruption detection support

fix: Configure MediaRecorder with consistent WebM/Opus format

fix: Implement correct WebSocket STT protocol and SSL handling

feat: Add comprehensive WebSocket STT debugging

chore: Bump version to v0.7.1

fix: Improve MCP fallback logic for STT service connection

feat: Implement unmute.sh streaming STT methodology v0.7.0

debug: Add audio debugging to trace 'Thank you very much' source v0.6.1

feat: Remove all demo simulation code - REAL STT ONLY v0.6.0

fix: Remove max_file_size parameter for HF Spaces compatibility v0.5.11

fix: Improve WebSocket stability and prevent spinning v0.5.10

feat: Add WebSocket STT integration with unmute.sh methodology v0.5.6

fix: Add max_file_size to resolve Gradio upload error v0.5.9

Disable demo TTS responses due to poor STT accuracy

fix: Add Hugging Face Spaces GPU support v0.5.8

fix: Add PyTorch dependencies and fix base64 imports v0.5.7

temp: Update WebSocket URLs to standard port 7860

feat: Add environment variable diagnostic script

feat: Add enhanced TTS connection error handling and diagnostics