"""WS 输出:裸 WebSocket 音频流(第二种输出方式)。

比 WebRTC 简单——没有 SDP/ICE/STUN/TURN,一条 WS 直接收发音频帧。
适合:服务端对接、话务网关、自定义客户端、调试。

约定:连接建立后,**第一条文本消息**发 JSON 启动参数:
    {"assistant_id": "asst_xxx"}            # 推荐:key 服务端解析
    {"inline_config": {...AssistantConfig}} # 调试:内联
之后的二进制消息即音频帧(protobuf,与 transports.py serializer 对应)。
"""

import json

from db.session import SessionLocal
from fastapi import APIRouter, WebSocket
from loguru import logger
from models import AssistantConfig
from services.config_resolver import resolve_runtime_config
from starlette.websockets import WebSocketDisconnect

# pipecat 重依赖,惰性导入(见 voice_webrtc.py 说明)

router = APIRouter()


async def _resolve_start_config(raw: str) -> AssistantConfig:
    data = json.loads(raw)
    if data.get("assistant_id"):
        async with SessionLocal() as session:
            return await resolve_runtime_config(session, data["assistant_id"])
    if data.get("inline_config"):
        return AssistantConfig(**data["inline_config"])
    raise ValueError("启动参数缺少 assistant_id 或 inline_config")


@router.websocket("/ws/stream")
async def voice_stream(websocket: WebSocket):
    from services.pipecat.pipeline import run_pipeline
    from services.pipecat.transports import build_ws_transport

    await websocket.accept()
    try:
        cfg = await _resolve_start_config(await websocket.receive_text())
        transport = build_ws_transport(websocket)
        # 直接 await:管线持续读这条 WS 的音频帧,直到对端断开
        await run_pipeline(transport, cfg)
    except WebSocketDisconnect:
        logger.info("WS 音频流断开")
    except Exception as e:
        logger.error(f"WS 音频流出错: {e}")