独立依赖

main.py

@@ -4,15 +4,13 @@ import asyncio
 import io
 import json
 import logging
-import os
 import time
 import wave
 from dataclasses import dataclass, field
 
 import numpy as np
-from aiortc import RTCPeerConnection, RTCSessionDescription
 from dotenv import load_dotenv
-from fastapi import FastAPI, Request, WebSocket, WebSocketDisconnect
+from fastapi import FastAPI, WebSocket, WebSocketDisconnect
 from fastapi.middleware.cors import CORSMiddleware
 from fastapi.responses import FileResponse, JSONResponse, StreamingResponse
 from fastapi.staticfiles import StaticFiles
@@ -26,11 +24,8 @@ from services.avatar import AvatarService
 from services.llm import LLMService
 from services.tts import TTSService
 from services.vad import VADService
-from webrtc.tracks import AudioBus, AvatarAudioTrack
 
 load_dotenv()
-if settings.hf_token:
-    os.environ["HF_TOKEN"] = settings.hf_token
 
 logging.basicConfig(level=logging.INFO)
 app = FastAPI(title="Visual Voice Chat")
@@ -50,8 +45,6 @@ tts_service = TTSService()
 vad_service = VADService()
 avatar_service = AvatarService()
 pipeline = ChatPipeline(arbitrator, asr_service, llm_service, tts_service)
-audio_bus = AudioBus(sample_rate=48000)
-pcs: set[RTCPeerConnection] = set()
 subtitle_clients: set[WebSocket] = set()
 animation_clients: set[WebSocket] = set()
 audio_clients: set[WebSocket] = set()
@@ -146,7 +139,6 @@ async def _broadcast_animation(payload: dict) -> None:
 def _runtime_payload() -> dict:
     return {
         "state": str(arbitrator.state),
-        "peers": len(pcs),
         "audio_clients": len(audio_clients),
         "subtitle_clients": len(subtitle_clients),
         "animation_clients": len(animation_clients),
@@ -341,7 +333,6 @@ async def _process_user_audio_chunk(mono_16k: np.ndarray, vad_buffer: np.ndarray
             barge_t0 = time.perf_counter()
             was_avatar_speaking = arbitrator.state == SessionState.AVATAR_SPEAKING
             await arbitrator.on_speech_start()
-            await audio_bus.clear()
             await _broadcast_audio_reset("speech-start")
             await _broadcast_animation(await avatar_service.build_reset_payload(reason="speech-start"))
             if was_avatar_speaking:
@@ -357,18 +348,6 @@ async def _process_user_audio_chunk(mono_16k: np.ndarray, vad_buffer: np.ndarray
     return vad_buffer
 
 
-async def _consume_user_audio(track) -> None:
-    vad_buffer = np.zeros(0, dtype=np.float32)
-    while True:
-        frame = await track.recv()
-        pcm = frame.to_ndarray()
-        sample_rate = getattr(frame, "sample_rate", 48000) or 48000
-        layout = getattr(frame, "layout", None)
-        channels = len(layout.channels) if layout and layout.channels else 1
-        mono_16k = _resample_to_16k_mono(pcm, sample_rate, channels)
-        vad_buffer = await _process_user_audio_chunk(mono_16k, vad_buffer)
-
-
 @app.get("/health")
 async def health():
     return _runtime_payload()
@@ -377,10 +356,9 @@ async def health():
 @app.get("/meta")
 async def meta():
     return {
-        "stun_url": settings.stun_url,
         "https_enabled": bool(settings.ssl_certfile and settings.ssl_keyfile),
-        "host": settings.webrtc_host,
-        "port": settings.webrtc_port,
+        "host": settings.http_host,
+        "port": settings.http_port,
         "avatar_protocol": settings.avatar_control_protocol,
     }
 
@@ -474,7 +452,6 @@ async def ws_audio(websocket: WebSocket):
                 elif event_type == "audio_reset":
                     _clear_buffered_audio()
                     runtime.pending_voice_turn = False
-                    await audio_bus.clear()
                     await _broadcast_audio_reset("client-reset")
                 elif event_type == "ping":
                     await websocket.send_text(json.dumps({"type": "pong"}, ensure_ascii=False))
@@ -567,7 +544,6 @@ async def chat_reset():
     runtime.vad_start_count = 0
     runtime.vad_end_count = 0
     runtime.last_animation_frame_count = 0
-    await audio_bus.clear()
     await _broadcast_audio_reset("chat-reset")
     await _broadcast_animation(await avatar_service.build_reset_payload(reason="chat-reset"))
     return {"ok": True}
@@ -583,60 +559,13 @@ async def root():
     return FileResponse("web/index.html")
 
 
-@app.post("/webrtc/offer")
-async def webrtc_offer(request: Request):
-    params = await request.json()
-    offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"])
-
-    replaced_previous = len(pcs) > 0
-    if replaced_previous:
-        old_peers = list(pcs)
-        await asyncio.gather(*(peer.close() for peer in old_peers), return_exceptions=True)
-        for peer in old_peers:
-            pcs.discard(peer)
-
-    pc = RTCPeerConnection()
-    pcs.add(pc)
-
-    @pc.on("connectionstatechange")
-    async def on_connectionstatechange():
-        if pc.connectionState in {"failed", "closed", "disconnected"}:
-            await pc.close()
-            pcs.discard(pc)
-
-    @pc.on("track")
-    def on_track(track):
-        if track.kind == "audio":
-            asyncio.create_task(_consume_user_audio(track))
-
-    pc.addTrack(AvatarAudioTrack(audio_bus=audio_bus))
-
-    await pc.setRemoteDescription(offer)
-    answer = await pc.createAnswer()
-    await pc.setLocalDescription(answer)
-
-    return JSONResponse(
-        {
-            "sdp": pc.localDescription.sdp,
-            "type": pc.localDescription.type,
-            "replaced_previous": replaced_previous,
-        }
-    )
-
-
-@app.on_event("shutdown")
-async def on_shutdown():
-    await asyncio.gather(*(pc.close() for pc in list(pcs)), return_exceptions=True)
-    pcs.clear()
-
-
 if __name__ == "__main__":
     import uvicorn
 
     uvicorn_kwargs = {
         "app": app,
-        "host": settings.webrtc_host,
-        "port": settings.webrtc_port,
+        "host": settings.http_host,
+        "port": settings.http_port,
     }
     if settings.ssl_certfile and settings.ssl_keyfile:
         uvicorn_kwargs["ssl_certfile"] = settings.ssl_certfile