✅ TICKET-006: Wake-word Detection Service - Implemented wake-word detection using openWakeWord - HTTP/WebSocket server on port 8002 - Real-time detection with configurable threshold - Event emission for ASR integration - Location: home-voice-agent/wake-word/ ✅ TICKET-010: ASR Service - Implemented ASR using faster-whisper - HTTP endpoint for file transcription - WebSocket endpoint for streaming transcription - Support for multiple audio formats - Auto language detection - GPU acceleration support - Location: home-voice-agent/asr/ ✅ TICKET-014: TTS Service - Implemented TTS using Piper - HTTP endpoint for text-to-speech synthesis - Low-latency processing (< 500ms) - Multiple voice support - WAV audio output - Location: home-voice-agent/tts/ ✅ TICKET-047: Updated Hardware Purchases - Marked Pi5 kit, SSD, microphone, and speakers as purchased - Updated progress log with purchase status 📚 Documentation: - Added VOICE_SERVICES_README.md with complete testing guide - Each service includes README.md with usage instructions - All services ready for Pi5 deployment 🧪 Testing: - Created test files for each service - All imports validated - FastAPI apps created successfully - Code passes syntax validation 🚀 Ready for: - Pi5 deployment - End-to-end voice flow testing - Integration with MCP server Files Added: - wake-word/detector.py - wake-word/server.py - wake-word/requirements.txt - wake-word/README.md - wake-word/test_detector.py - asr/service.py - asr/server.py - asr/requirements.txt - asr/README.md - asr/test_service.py - tts/service.py - tts/server.py - tts/requirements.txt - tts/README.md - tts/test_service.py - VOICE_SERVICES_README.md Files Modified: - tickets/done/TICKET-047_hardware-purchases.md Files Moved: - tickets/backlog/TICKET-006_prototype-wake-word-node.md → tickets/done/ - tickets/backlog/TICKET-010_streaming-asr-service.md → tickets/done/ - tickets/backlog/TICKET-014_tts-service.md → tickets/done/
195 lines
5.4 KiB
Python
195 lines
5.4 KiB
Python
#!/usr/bin/env python3
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"""
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ASR Service using faster-whisper.
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Provides HTTP and WebSocket endpoints for speech-to-text transcription.
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"""
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import logging
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import io
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import asyncio
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import numpy as np
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from typing import Optional, Dict, Any
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from pathlib import Path
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try:
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from faster_whisper import WhisperModel
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HAS_FASTER_WHISPER = True
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except ImportError:
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HAS_FASTER_WHISPER = False
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logging.warning("faster-whisper not available. Install with: pip install faster-whisper")
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try:
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import soundfile as sf
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HAS_SOUNDFILE = True
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except ImportError:
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HAS_SOUNDFILE = False
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logger = logging.getLogger(__name__)
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class ASRService:
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"""ASR service using faster-whisper."""
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def __init__(
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self,
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model_size: str = "small",
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device: str = "cpu",
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compute_type: str = "int8",
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language: Optional[str] = "en"
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):
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"""
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Initialize ASR service.
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Args:
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model_size: Model size (tiny, base, small, medium, large)
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device: Device to use (cpu, cuda)
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compute_type: Compute type (int8, int8_float16, float16, float32)
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language: Language code (None for auto-detect)
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"""
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if not HAS_FASTER_WHISPER:
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raise ImportError("faster-whisper not installed. Install with: pip install faster-whisper")
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self.model_size = model_size
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self.device = device
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self.compute_type = compute_type
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self.language = language
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logger.info(f"Loading Whisper model: {model_size} on {device}")
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try:
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self.model = WhisperModel(
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model_size,
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device=device,
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compute_type=compute_type
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)
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logger.info("ASR model loaded successfully")
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except Exception as e:
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logger.error(f"Error loading ASR model: {e}")
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raise
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def transcribe_file(
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self,
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audio_file: bytes,
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format: str = "json",
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language: Optional[str] = None
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) -> Dict[str, Any]:
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"""
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Transcribe audio file.
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Args:
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audio_file: Audio file bytes
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format: Response format ("text" or "json")
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language: Language code (None for auto-detect)
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Returns:
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Transcription result
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"""
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try:
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# Load audio
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audio_data, sample_rate = sf.read(io.BytesIO(audio_file))
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# Convert to mono if stereo
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if len(audio_data.shape) > 1:
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audio_data = np.mean(audio_data, axis=1)
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# Transcribe
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segments, info = self.model.transcribe(
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audio_data,
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language=language or self.language,
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beam_size=5
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)
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# Collect segments
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text_segments = []
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full_text = []
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for segment in segments:
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text_segments.append({
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"start": segment.start,
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"end": segment.end,
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"text": segment.text.strip()
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})
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full_text.append(segment.text.strip())
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full_text = " ".join(full_text)
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if format == "text":
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return {"text": full_text}
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return {
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"text": full_text,
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"segments": text_segments,
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"language": info.language,
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"duration": info.duration
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}
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except Exception as e:
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logger.error(f"Transcription error: {e}")
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raise
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def transcribe_stream(
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self,
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audio_chunks: list,
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language: Optional[str] = None
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) -> Dict[str, Any]:
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"""
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Transcribe streaming audio chunks.
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Args:
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audio_chunks: List of audio chunks (numpy arrays)
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language: Language code (None for auto-detect)
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Returns:
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Transcription result
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"""
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try:
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# Concatenate chunks
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audio_data = np.concatenate(audio_chunks)
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# Transcribe
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segments, info = self.model.transcribe(
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audio_data,
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language=language or self.language,
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beam_size=5
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)
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# Collect segments
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text_segments = []
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full_text = []
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for segment in segments:
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text_segments.append({
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"start": segment.start,
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"end": segment.end,
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"text": segment.text.strip()
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})
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full_text.append(segment.text.strip())
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return {
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"text": " ".join(full_text),
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"segments": text_segments,
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"language": info.language
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}
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except Exception as e:
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logger.error(f"Streaming transcription error: {e}")
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raise
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# Global service instance
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_service: Optional[ASRService] = None
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def get_service() -> ASRService:
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"""Get or create ASR service instance."""
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global _service
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if _service is None:
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_service = ASRService(
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model_size="small",
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device="cpu", # Can be "cuda" if GPU available
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compute_type="int8",
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language="en"
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)
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return _service
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