Implement WebSocket real-time audio streaming for Hermes gateway

This commit is contained in:
Adolfo Reyna
2026-06-19 13:58:42 -04:00
parent b2bcdb5b79
commit b88cb34196
4 changed files with 935 additions and 0 deletions
+253
View File
@@ -652,7 +652,260 @@ class HermesESP32VoiceGatewayMixin:
},
)
async def _handle_esp32_voice_websocket(self, request: Any) -> Any:
"""GET /api/esp32/voice/ws — real-time WebSocket audio streaming and status route."""
if web is None:
raise RuntimeError("aiohttp is required for the ESP32 voice websocket endpoint")
# 1. Check authentication first (HTTP headers)
auth_err = self._check_auth(request)
if auth_err:
return auth_err
# Prepare websocket response
ws = web.WebSocketResponse()
await ws.prepare(request)
device_id = (request.headers.get("X-Device-ID") or request.query.get("device_id") or "default").strip()[:80] or "default"
# 2. Receive JSON start frame
try:
msg = await asyncio.wait_for(ws.receive(), timeout=10)
if msg.type != web.WSMsgType.TEXT:
await ws.send_json({"event": "error", "message": "Expected text start frame", "code": "protocol_error"})
await ws.close()
return ws
data = json.loads(msg.data)
if data.get("event") != "start":
await ws.send_json({"event": "error", "message": "Expected start event", "code": "protocol_error"})
await ws.close()
return ws
if "device_id" in data:
device_id = str(data["device_id"]).strip()[:80] or device_id
except asyncio.TimeoutError:
await ws.close()
return ws
except Exception as exc:
try:
await ws.send_json({"event": "error", "message": f"Invalid start payload: {exc}", "code": "protocol_error"})
except Exception:
pass
await ws.close()
return ws
# Send ready status
await ws.send_json({"event": "ready"})
await ws.send_json({"event": "listening"})
try:
from hermes_constants import get_hermes_dir
audio_dir = get_hermes_dir("cache/audio", "audio_cache") / "esp32"
except Exception:
audio_dir = Path.home() / ".hermes" / "cache" / "audio" / "esp32"
audio_dir.mkdir(parents=True, exist_ok=True)
safe_device = re.sub(r"[^A-Za-z0-9_.-]+", "_", device_id)[:80] or "default"
raw_path = audio_dir / f"ws_{safe_device}_{int(time.time() * 1000)}_{uuid.uuid4().hex[:8]}.s16le"
total_bytes = 0
start_time = time.time()
last_frame_time = time.time()
cancelled = False
timings = {"ws_accepted": start_time}
try:
with open(raw_path, "wb") as f:
while True:
now = time.time()
elapsed = now - last_frame_time
if elapsed > 10.0:
await ws.send_json({"event": "error", "message": "Idle timeout exceeded", "code": "timeout"})
break
if now - start_time > 60.0:
await ws.send_json({"event": "error", "message": "Max session duration exceeded", "code": "duration_limit"})
break
timeout_left = min(10.0 - elapsed, 60.0 - (now - start_time))
if timeout_left <= 0:
break
try:
msg = await asyncio.wait_for(ws.receive(), timeout=timeout_left)
except asyncio.TimeoutError:
await ws.send_json({"event": "error", "message": "Idle timeout exceeded", "code": "timeout"})
break
last_frame_time = time.time()
if msg.type == web.WSMsgType.BINARY:
if total_bytes + len(msg.data) > 8_000_000:
await ws.send_json({"event": "error", "message": "Max audio size exceeded", "code": "size_limit"})
break
f.write(msg.data)
total_bytes += len(msg.data)
if "first_audio" not in timings:
timings["first_audio"] = last_frame_time
elif msg.type == web.WSMsgType.TEXT:
try:
data = json.loads(msg.data)
event = data.get("event")
if event == "stop":
timings["stop_received"] = last_frame_time
break
elif event == "cancel":
cancelled = True
break
except Exception:
pass
elif msg.type in (web.WSMsgType.CLOSE, web.WSMsgType.CLOSING):
break
elif msg.type == web.WSMsgType.ERROR:
break
except Exception as exc:
logger.exception("Error in websocket audio loop")
try:
await ws.send_json({"event": "error", "message": f"Internal server error: {exc}", "code": "server_error"})
except Exception:
pass
raw_path.unlink(missing_ok=True)
await ws.close()
return ws
if cancelled or ws.closed or total_bytes < 3200 or "stop_received" not in timings:
raw_path.unlink(missing_ok=True)
if not ws.closed:
if total_bytes < 3200:
await ws.send_json({"event": "error", "message": "Audio recording too short", "code": "audio_too_short"})
else:
await ws.send_json({"event": "error", "message": "Session cancelled", "code": "cancelled"})
await ws.close()
return ws
wav_path = raw_path.with_suffix(".wav")
try:
with wave.open(str(wav_path), "wb") as wav:
wav.setnchannels(1)
wav.setsampwidth(2)
wav.setframerate(16000)
wav.writeframes(raw_path.read_bytes())
finally:
raw_path.unlink(missing_ok=True)
timings["wav_finalized"] = time.time()
try:
from tools.transcription_tools import transcribe_audio
stt_result = await asyncio.wait_for(
asyncio.to_thread(transcribe_audio, str(wav_path)),
timeout=45
)
timings["stt_done"] = time.time()
if not stt_result.get("success"):
await ws.send_json({"event": "error", "message": stt_result.get("error") or "Transcription failed", "code": "stt_failed"})
await ws.close()
return ws
transcript = str(stt_result.get("transcript") or "").strip()
if not transcript:
await ws.send_json({"event": "error", "message": "Transcription empty", "code": "empty_transcript"})
await ws.close()
return ws
await ws.send_json({"event": "transcript", "text": transcript})
await ws.send_json({"event": "thinking"})
response_data, final_text = await self._run_esp32_voice_turn(
transcript=transcript,
device_id=device_id,
instructions=request.headers.get("X-Hermes-Instructions") or None,
)
timings["agent_done"] = time.time()
await ws.send_json({"event": "response_text", "text": final_text})
from tools.tts_tool import text_to_speech_tool
tts_raw = await asyncio.to_thread(text_to_speech_tool, final_text)
tts_result = json.loads(tts_raw)
timings["tts_done"] = time.time()
if not tts_result.get("success"):
await ws.send_json({"event": "error", "message": tts_result.get("error") or "TTS generation failed", "code": "tts_failed"})
await ws.close()
return ws
tts_path = str(tts_result.get("file_path") or "")
if not tts_path or not Path(tts_path).exists():
await ws.send_json({"event": "error", "message": "TTS file missing", "code": "tts_file_missing"})
await ws.close()
return ws
response_audio_path = self._convert_audio_to_esp32_wav(
tts_path,
output_dir=audio_dir,
device_id=device_id
)
wav_bytes = response_audio_path.read_bytes()
await ws.send_json({
"event": "audio_start",
"content_type": "audio/wav",
"bytes": len(wav_bytes)
})
pos = 0
chunk_size = 4096
while pos < len(wav_bytes):
chunk = wav_bytes[pos : pos + chunk_size]
await ws.send_bytes(chunk)
pos += chunk_size
await ws.send_json({"event": "audio_end"})
await ws.send_json({"event": "done"})
timings["audio_sent"] = time.time()
except Exception as exc:
logger.exception("Error processing voice session")
try:
await ws.send_json({"event": "error", "message": f"Processing error: {exc}", "code": "processing_failed"})
except Exception:
pass
finally:
wav_path.unlink(missing_ok=True)
try:
await ws.close()
except Exception:
pass
if "audio_sent" in timings:
duration = timings["audio_sent"] - timings["ws_accepted"]
status = "completed"
else:
duration = time.time() - timings["ws_accepted"]
status = "failed"
logger.info("ESP32 Voice WebSocket Session Completed. Timings: %s", timings)
audit_event = {
"timestamp": time.time(),
"device_id": device_id,
"source_ip": getattr(request, "remote", "") or "",
"duration": duration,
"byte_count": total_bytes,
"timings": timings,
"status": status
}
self._append_esp32_voice_audit(audio_dir, audit_event)
return ws
def register_esp32_voice_route(app: Any, api_server: HermesESP32VoiceGatewayMixin) -> None:
"""Register the endpoint on an aiohttp app."""
app.router.add_post("/api/esp32/voice", api_server._handle_esp32_voice)
app.router.add_get("/api/esp32/voice/ws", api_server._handle_esp32_voice_websocket)