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fix: Catch buffer size error in MCP audio handler v0.4.7β
Peter Michael Gits +1: Claude- Fix persistent "buffer size must be a multiple of element size" error - Add specific ValueError handling for numpy buffer conversion issues - Provide graceful fallback when audio buffer format is incompatible - Return meaningful WebRTC fallback message instead of crashing - Maintain audio duration calculation for demo responses - Error occurs in Gradio fallback handler even when using WebRTC Buffer Error Handling: - Catch ValueError during audio_array.astype(np.int16) conversion - Detect specific buffer size error message - Return WebRTC fallback response with duration info - Prevent crashes during audio processing π€ Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>3863173 -
feat: WebRTC-first implementation with demo audio processing v0.4.4β
Peter Michael Gits +1: Claude- Comment out failing HTTP STT calls (preserve for future) - Enable WebRTC FastAPI app as primary deployment - Implement WebRTC audio processing with characteristics detection - Remove WSGI middleware complexity - direct FastAPI deployment - Create functional WebSocket endpoints for real-time audio - Add audio duration and sample rate analysis in WebRTC demo - Fallback to Gradio-only if WebRTC fails WebRTC Endpoints Available: - /ws/webrtc/{client_id} - WebSocket for audio streaming - /webrtc/demo - Interactive demo with microphone - /webrtc/test - API status check Note: HTTP STT services preserved in comments for later integration π€ Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>65f90da -
feat: WebRTC Phase 1 - Remove demo modes and create WebRTC architecture v0.4.0β
Peter Michael Gits +1: ClaudeBREAKING CHANGE: Eliminated all fake/demo responses - real services only β Demo Mode Removal: - Force demo_mode = False in MCPAudioHandler initialization - Replace _initialize_mcp_services with _initialize_real_services - Real STT/TTS service calls only - errors instead of fake responses β WebRTC Architecture Created: - Isolated webrtc/ directory structure - client/, server/, utils/ modules - Complete architecture documentation - Safe dependency additions (websockets, sounddevice, webrtcvad) β No Module Collisions: - Separate WebRTC namespace - FastAPI WebSocket integration - Compatible with existing async/uvicorn setup Next: Phase 1 WebSocket + real-time audio streaming implementation π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>66471b4 -
fix: Comprehensive HTTP API approach to bypass Gradio client issues v0.3.42β
Peter Michael Gits +1: Claude- Try direct HTTP API calls with multiple endpoint patterns - Test /api/predict, /call/predict, /api/transcribe_audio, /call/transcribe_audio - Multiple payload formats for maximum compatibility - Detailed response debugging and JSON parsing - Fallback to Gradio client only if direct API fails π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>6dac05e -
debug: Add detailed Gradio client debugging for STT connection v0.3.41β
Peter Michael Gits +1: Claude- Add logging at each step of Gradio client creation - Show raw results from STT service API calls - Debug 'Expecting value: line 1 column 1' JSON parsing error - Isolate whether issue is client creation or API calls π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>0d31e42 -
fix: Try multiple Gradio API endpoints to connect to STT service v0.3.40β
Peter Michael Gits +1: Claude- Try default endpoint (None), /predict, /transcribe, and index 0 - Add detailed logging for each endpoint attempt - Systematic approach to find working Gradio API connection - Should resolve 404 errors and connect to actual STT transcription π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>d1d27c3 -
fix: Correct HTTP STT API call format and prevent demo fallback v0.3.39β
Peter Michael Gits +1: Claude- Fix Gradio client call to extract transcription correctly from (status, transcription, timestamps) - Replace demo text fallbacks with proper error messages - Add detailed HTTP response debugging - Prevent demo mode when HTTP fails with 404 π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>ec9272d
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fix: Enable HTTP STT service calls to bypass demo mode v0.3.38β
Peter Michael Gits +1: Claude- Fix HTTP fallback to properly exit demo mode when STT service available - Prioritize HTTP STT calls over demo simulation - Audio processing now uses real STT service instead of hardcoded responses - Fixes: 'Hi, I'm testing the MCP voice input' demo response issue π Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>fd0d36e -
feat: Complete MCP integration with comprehensive logging fixes v0.3.32β
Peter Michael Gits +1: Claudeπ Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>9d0b9f5 -
feat: Implement MCP-based audio handler for STT/TTS services v0.3.30β
Peter Michael Gits +1: Claude- Add MCPAudioHandler to replace HTTP-based audio processing - Create MCP service discovery and connection framework - Add MCP client dependency for reliable service integration - Implement audio processing pipeline optimized for MCP services - Add enhanced debugging for MCP service interactions - Maintain backward compatibility with demo mode for testing This provides cleaner integration than HTTP endpoints and better reliability. π€ Generated with [Claude Code](https://claude.ai/code) Co-Authored-By: Claude <noreply@anthropic.com>c045c6a