jarvis-cloud / backend /voice /audio_ws.py
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# backend/voice/audio_ws.py
import asyncio
import json
import logging
import numpy as np
import websockets
import wave
import io
import os
from datetime import datetime
from .vad import SileroVAD
from .stt import STTPipeline
from .tts import TTSPipeline
from .wake_word import WakeWordDetector
from backend.ws.agent_ws import ws_manager
import sys
parent_dir = os.path.dirname(os.path.dirname(os.path.dirname(os.path.abspath(__file__))))
if parent_dir not in sys.path:
sys.path.insert(0, parent_dir)
from modules.assistant_identity import set_mode
# Shared pipeline lock — also used by space_injection.py so wake-word utterances
# and space-ack speech cannot physically overlap.
from backend.voice.space_injection import get_pipeline_lock
AUDIO_LOGS_DIR = os.path.join(os.path.dirname(__file__), "audio_logs")
os.makedirs(AUDIO_LOGS_DIR, exist_ok=True)
TRANSCRIPT_LOG_PATH = os.path.join(AUDIO_LOGS_DIR, "transcripts_log.txt")
active_audio_ws = None
class AudioWebSocketServer:
def __init__(self, host="127.0.0.1", port=8767):
global active_audio_ws
active_audio_ws = self
self.host = host
self.port = port
self.vad = SileroVAD()
self.stt = STTPipeline()
self.tts = TTSPipeline()
self.wake_word = WakeWordDetector()
self.stop_future = None
self.interrupt_flag = False
# Use the shared pipeline lock so space_injection cannot fire mid-utterance
self.wake_lock = get_pipeline_lock()
def interrupt(self):
self.interrupt_flag = True
async def handler(self, websocket):
logging.info("Client connected to Voice WebSocket.")
speech_buffer = bytearray()
is_speaking = False
silence_chunks = 0
MAX_SILENCE_CHUNKS = 15 # Approx 0.5s of silence to trigger end of utterance
is_tts_playing = False
try:
async for message in websocket:
if self.interrupt_flag:
speech_buffer.clear()
is_speaking = False
silence_chunks = 0
self.interrupt_flag = False
if isinstance(message, bytes):
if is_tts_playing:
continue # Mute mic input while TTS is speaking to prevent self-triggering
audio_np = np.frombuffer(message, dtype=np.int16).astype(np.float32) / 32768.0
# 1. Wake Word Detection (openWakeWord)
keyword = self.wake_word.process_frame(audio_np)
if keyword:
async with self.wake_lock:
if "friday" in keyword:
set_mode("friday")
elif "jarvis" in keyword:
set_mode("jarvis")
logging.info(f"openWakeWord triggered: {keyword}")
await ws_manager.broadcast({"event": "voice:wake_word", "payload": {"agent": keyword}})
speech_buffer = bytearray() # Clear buffer to start listening
is_speaking = True
silence_chunks = 0
continue
# 2. VAD Confirmation
vad_res = self.vad.process_chunk(audio_np)
if vad_res:
is_speaking = True
silence_chunks = 0
speech_buffer.extend(message)
elif is_speaking:
speech_buffer.extend(message)
silence_chunks += 1
if silence_chunks >= MAX_SILENCE_CHUNKS:
is_speaking = False
silence_chunks = 0
if len(speech_buffer) > 16000:
logging.info("Utterance complete. Transcribing...")
wav_io = io.BytesIO()
with wave.open(wav_io, 'wb') as wf:
wf.setnchannels(1)
wf.setsampwidth(2)
wf.setframerate(16000)
wf.writeframes(speech_buffer)
# 3. Whisper Transcription
text, detected_lang = await self.stt.transcribe(wav_io.getvalue())
if text.strip():
timestamp = datetime.now().strftime("%Y-%m-%d_%H-%M-%S")
wav_filepath = os.path.join(AUDIO_LOGS_DIR, f"{timestamp}.wav")
with open(wav_filepath, "wb") as f:
f.write(wav_io.getvalue())
await websocket.send(json.dumps({
"type": "transcript",
"text": text.strip(),
"language": detected_lang
}))
# --- JARVIS 10X Gaming Coach Interrupt ---
# If the user yells a tactical interrupt, kill the audio immediately.
interrupt_phrases = ["quiet", "stop", "shut up", "enemy on me", "contact"]
if any(phrase in text.lower() for phrase in interrupt_phrases):
logging.warning(f"JARVIS 10X Tactical Interrupt Detected: '{text.strip()}'")
# Send hard kill signal to frontend audio player
await websocket.send(json.dumps({"type": "audio_interrupt"}))
# Skip routing to LLM for this utterance unless it's a query
if "what went wrong" not in text.lower():
speech_buffer = bytearray()
continue
# 4. Route to LLM
logging.info(f"Routing to ReAct LLM Agent: {text.strip()} (Lang: {detected_lang})")
await ws_manager.handle_client_event(
websocket=None,
data={"type": "chat", "text": text.strip(), "language": detected_lang}
)
speech_buffer = bytearray()
elif isinstance(message, str):
try:
data = json.loads(message)
msg_type = data.get("type")
if msg_type == "tts_request":
text = data.get("text", "")
personality = data.get("personality", "jarvis")
language = data.get("language", "en")
if text:
logging.info(f"Synthesizing TTS for: {text} in {language}")
wav_bytes = await self.tts.synthesize(text, personality, language)
async def _play_and_mute():
nonlocal is_tts_playing
is_tts_playing = True
await websocket.send(json.dumps({
"type": "tts_response_start",
"text": text
}))
await websocket.send(wav_bytes)
await websocket.send(json.dumps({
"type": "tts_response_end"
}))
# 16-bit PCM = 2 bytes per sample @ 24kHz
audio_duration = len(wav_bytes) / 24000.0 / 2.0
await asyncio.sleep(audio_duration + 0.5) # +0.5s trailing silence buffer
is_tts_playing = False
asyncio.create_task(_play_and_mute())
except json.JSONDecodeError:
pass
except websockets.exceptions.ConnectionClosed:
logging.info("Client disconnected from Voice WebSocket.")
except Exception as e:
logging.error(f"Voice WebSocket Error: {e}")
async def start_server(self):
self.stop_future = asyncio.Future()
server = None
for p in range(self.port, self.port + 11):
try:
server = await websockets.serve(self.handler, self.host, p)
self.port = p
logging.info(f"Voice WebSocket running on ws://{self.host}:{self.port}")
break
except OSError as e:
if "address already in use" in str(e).lower() or "10048" in str(e):
logging.warning(f"Voice WS Port {p} in use, trying next...")
continue
raise
if server:
await self.stop_future # run until stopped
server.close()
await server.wait_closed()
logging.info("Voice WebSocket server shut down cleanly.")
def stop(self):
if self.stop_future and not self.stop_future.done():
self.stop_future.set_result(True)
def run_in_thread(host="127.0.0.1", port=8767):
import os
if os.environ.get("CLOUD_ENV", "false").lower() == "true":
logging.info("[AudioWS] Cloud mode — skipping local audio WebSocket server (no mic on HF Spaces).")
return None
server = AudioWebSocketServer(host=host, port=port)
def _run():
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
loop.run_until_complete(server.start_server())
import threading
t = threading.Thread(target=_run, daemon=True)
t.start()
return server
if __name__ == "__main__":
logging.basicConfig(level=logging.INFO)
asyncio.run(AudioWebSocketServer().start_server())