UNMUTE.SH METHODOLOGY IMPLEMENTATION:
- Buffer audio chunks during recording (like unmute.sh streaming)
- Apply flush trick when recording stops - combine all chunks into complete audio
- Process complete audio with is_final=True for proper transcription finalization
- Clear buffer after processing to prevent chunk contamination
- Add flush trick markers and metadata following unmute.sh patterns
Key Changes:
✅ Buffer chunks during recording instead of processing individually
✅ Implement flush trick: combine chunks + process with is_final=True
✅ Send only final transcription after complete audio processing
✅ Clear buffers after each recording session
✅ Add unmute.sh-style metadata (is_final, chunks_processed)
This should fix the 'Thank you very much' issue by properly implementing
unmute.sh's streaming methodology with flush trick for clean transcriptions.
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
fix: Add PyTorch dependencies and fix base64 imports v0.5.7
Peter Michael Gits+1: Claude
e985487
- Add missing PyTorch, TorchAudio, and Transformers dependencies to requirements.txt
- Fix missing base64 import statements in WebSocket handler
- Add better HTTP 503 error handling for cold starts
- Update to version 0.5.7 with comprehensive fixes
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
temp: Update WebSocket URLs to standard port 7860
Peter Michael Gits+1: Claude
22227a1
- Revert STT back to port 7860 (services now forced to WebSocket mode)
- Both STT and TTS should run WebSocket endpoints on port 7860
- Temporary fix while debugging environment variable detection
- Should resolve TTS connection issues
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
feat: Add enhanced TTS connection error handling and diagnostics
Peter Michael Gits+1: Claude
57bc78b
- Add timeout handling for TTS WebSocket connections
- Improve error messages with specific diagnostic hints
- Detect when TTS service is in wrong mode (Gradio vs WebSocket)
- Guide users to set TTS_SERVICE_MODE=websocket environment variable
- Better debugging for TTS connection failures
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
fix: Update STT WebSocket URL to use port 7861
Peter Michael Gits+1: Claude
f118868
- STT service is running in triple mode with WebSocket on port 7861
- Update VoiceCal to connect to correct STT WebSocket endpoint
- Fixes STT connection issues while TTS service configuration is pending
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
feat: Deploy complete VoiceCal application with all files v0.5.6
Peter Michael Gits+1: Claude
5e8a657
- Add all application files: app.py, requirements.txt, core modules
- Include WebRTC integration and voice service components
- Complete Gradio application ready for Hugging Face deployment
- Comprehensive README with VoiceCal branding and documentation
- All dependencies and Docker configuration included
🤖 Generated with [Claude Code](https://claude.ai/code)
Co-Authored-By: Claude <noreply@anthropic.com>
