# Ticket: Implement Streaming Audio Capture → ASR Service

## Ticket Information

- **ID**: TICKET-010
- **Title**: Implement Streaming Audio Capture → ASR Service
- **Type**: Feature
- **Priority**: High
- **Status**: Backlog
- **Track**: Voice I/O
- **Milestone**: Milestone 2 - Voice Chat MVP
- **Created**: 2024-01-XX

## Description

Build streaming ASR service:
- Implement audio capture (GStreamer/ffmpeg or browser getUserMedia)
- Create WebSocket endpoint for audio streaming
- Integrate selected ASR engine
- Stream audio chunks to ASR and return text segments
- Handle start/stop based on wake-word events

## Acceptance Criteria

- [ ] Audio capture working (from mic or WebSocket)
- [ ] WebSocket endpoint for audio streaming
- [ ] ASR engine integrated
- [ ] Text segments returned with timestamps
- [ ] Handles wake-word start/stop events
- [ ] Streaming latency acceptable (< 2s end-to-end)

## Technical Details

Implementation:
- Audio capture: PyAudio, GStreamer, or browser MediaRecorder
- WebSocket server for real-time streaming
- ASR processing: faster-whisper or selected engine
- Return format: JSON with text, timestamps, confidence

## Dependencies

- TICKET-009 (ASR engine selection)
- Wake-word event flow defined (from TICKET-006)

## Related Files

- `home-voice-agent/asr/` (to be created)

## Notes

Can run in parallel with TTS and LLM work. Needs wake-word integration for start/stop.