059ad8162e
- Introduce new database models for conversation sessions, messages, and artifacts to support conversation history tracking. - Implement API routes for listing conversations and retrieving detailed conversation data, enhancing user interaction with historical records. - Add a conversation recorder service to persist conversation messages in real-time without disrupting ongoing calls. - Update the frontend to display conversation history, including filtering and sorting options, improving user experience. - Enhance the pipeline to integrate conversation history recording seamlessly during interactions.
57 lines
2.1 KiB
Python
57 lines
2.1 KiB
Python
"""WS 输出:裸 WebSocket 音频流(第二种输出方式)。
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比 WebRTC 简单——没有 SDP/ICE/STUN/TURN,一条 WS 直接收发音频帧。
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适合:服务端对接、话务网关、自定义客户端、调试。
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约定:连接建立后,**第一条文本消息**发 JSON 启动参数:
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{"assistant_id": "asst_xxx"} # 推荐:key 服务端解析
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{"inline_config": {...AssistantConfig}} # 调试:内联
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之后的二进制消息即音频帧(protobuf,与 transports.py serializer 对应)。
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"""
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import json
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from db.session import SessionLocal
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from fastapi import APIRouter, WebSocket
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from loguru import logger
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from models import AssistantConfig
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from services.auth import require_admin_websocket
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from services.config_resolver import resolve_runtime_config
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from starlette.websockets import WebSocketDisconnect
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# pipecat 重依赖,惰性导入(见 voice_webrtc.py 说明)
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router = APIRouter()
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async def _resolve_start_config(raw: str) -> tuple[AssistantConfig, str | None]:
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data = json.loads(raw)
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if data.get("assistant_id"):
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async with SessionLocal() as session:
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return (
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await resolve_runtime_config(session, data["assistant_id"]),
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data["assistant_id"],
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)
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if data.get("inline_config"):
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return AssistantConfig(**data["inline_config"]), None
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raise ValueError("启动参数缺少 assistant_id 或 inline_config")
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@router.websocket("/ws/stream")
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async def voice_stream(websocket: WebSocket):
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from services.pipecat.pipeline import run_pipeline
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from services.pipecat.transports import build_ws_transport
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if not await require_admin_websocket(websocket):
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return
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await websocket.accept()
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try:
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cfg, assistant_id = await _resolve_start_config(await websocket.receive_text())
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transport = build_ws_transport(websocket)
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# 直接 await:管线持续读这条 WS 的音频帧,直到对端断开
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await run_pipeline(transport, cfg, assistant_id=assistant_id, channel="websocket")
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except WebSocketDisconnect:
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logger.info("WS 音频流断开")
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except Exception as e:
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logger.error(f"WS 音频流出错: {e}")
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