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a31f556 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 | # 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())
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