656 lines
26 KiB
Python
656 lines
26 KiB
Python
#!/usr/bin/env python3
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"""Pipecat SmallWebRTC 服务端（实验）。
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**目的**：验证 Pipecat 传输层与自有引擎能否协同工作，并拿到真实传输下的延迟数字。
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不是产品代码——见文件末尾"这还不是什么"。
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结构：
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┌──────────────── Pipecat ────────────────┐
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│ transport.input() → EngineProcessor │
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│ ↓ │
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│ transport.output() │
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└─────────────────────────────────────────┘
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↑
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我们的 ConversationEngine
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（STT→记忆→LLM→TTS 全在里面）
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EngineProcessor 把整条管线"吞"进一个处理器——这是 A 方案的核心假设：
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**Pipecat 提供传输与帧模型，会话语义归我们的引擎。** 这个实验就是在检验它成不成立。
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用法：
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# 服务器上
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.venv/bin/python services/bot_server.py --host 0.0.0.0 --port 7860
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# 浏览器打开 http://<服务器IP>:7860/ 点连接后说话
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"""
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from __future__ import annotations
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import argparse
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import asyncio
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import logging
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import os
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import sys
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import time
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from collections import deque
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from datetime import UTC, datetime
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from pathlib import Path
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import numpy as np
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import pipecat.processors.frameworks.rtvi.models as RTVI
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sys.path.insert(0, str(Path(__file__).resolve().parent.parent))
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from fastapi import FastAPI
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from fastapi.responses import RedirectResponse
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from pipecat.frames.frames import (
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CancelFrame,
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EndFrame,
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InputAudioRawFrame,
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InterruptionFrame,
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LLMConfigureOutputFrame,
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LLMContextFrame,
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LLMFullResponseEndFrame,
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LLMFullResponseStartFrame,
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LLMMessagesAppendFrame,
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LLMTextFrame,
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OutputAudioRawFrame,
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StartFrame,
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TranscriptionFrame,
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)
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from pipecat.pipeline.pipeline import Pipeline
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from pipecat.pipeline.worker import PipelineParams, PipelineWorker
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from pipecat.processors.aggregators.llm_context import LLMContext
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from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
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from pipecat.processors.frameworks.rtvi import (
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RTVIObserverParams,
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RTVIProcessor,
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)
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from pipecat.transports.base_transport import TransportParams
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from pipecat.transports.smallwebrtc.request_handler import (
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SmallWebRTCRequest,
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SmallWebRTCRequestHandler,
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)
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from pipecat.transports.smallwebrtc.transport import SmallWebRTCTransport
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from pipecat.utils.asyncio.task_manager import TaskManager
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from pipecat.workers.base_worker import WorkerParams
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from app.config import AppConfig
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from app.engine import ConversationEngine, EventKind
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from app.identity import ManualIdentityProvider
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from app.llm import FallbackProvider, create_provider
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from app.memory import SQLiteMemoryStore
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from app.voice import StreamingSTT, create_tts
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logging.basicConfig(level=logging.INFO,
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format="%(asctime)s %(levelname)s %(message)s")
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log = logging.getLogger("bot")
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# Pipecat 会在每个 worker 启动时预热 NLTK 分句器。固定到私有、可预装的目录，
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# 避免使用权限过宽的 ~/nltk_data，也避免首个会话临时联网下载。
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DEFAULT_NLTK_DATA = Path.home() / ".local/share/ai-companion/nltk_data"
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os.environ.setdefault("NLTK_DATA", str(DEFAULT_NLTK_DATA))
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#: 输入音频格式（WebRTC 侧协商后的格式）
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IN_RATE = 16000
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IN_CHANNELS = 1
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#: 输出音频格式（Qwen3-TTS 输出 24kHz）
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OUT_RATE = 24000
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#: 管线参数。给 PipelineWorker 构造用。
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PIPELINE_PARAMS = PipelineParams(
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audio_in_sample_rate=IN_RATE,
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audio_out_sample_rate=OUT_RATE,
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enable_metrics=True,
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)
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#: 注意：PipelineWorker.run() 要的是 WorkerParams（内含 task_manager），
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#: **不是** PipelineParams。传错会依次报两个错，且都发生在后台任务里，
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#: 主日志里什么都看不到：
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#: 1. 不传 → "PipelineWorker.run() missing 1 required positional argument: 'params'"
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#: 2. 传错 → "'PipelineParams' object has no attribute 'task_manager'"
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#: 而且 TaskManager() 内部调用 asyncio.get_running_loop()，
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#: **不能在模块级创建**（会报 "no running event loop"），必须在异步上下文里建。
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#: 参考实现：pipecat/workers/runner.py
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# ---------------------------------------------------------------- 能量 VAD
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class SimpleVAD:
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"""极简能量判定。
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**这是实验用的占位实现**，不是最终方案。方案 §5 把"说完判定"列为需要
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实测的部分，正式的 VAD 还没选（候选：Pipecat 的 Silero、
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sherpa-onnx VAD、或客户端 VAD + 服务端二次确认）。
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用它的唯一原因是：先把传输层跑通，VAD 的好坏不影响"Pipecat 能不能
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当传输层用"这个结论。
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"""
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def __init__(self, *, threshold: float = 350.0,
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silence_ms: int = 700, min_speech_ms: int = 300):
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self.threshold = threshold # PCM16 的 RMS 阈值
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self.silence_ms = silence_ms
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self.min_speech_ms = min_speech_ms
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self._silence = 0.0
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self._speech = 0.0
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def reset(self) -> None:
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self._silence = 0.0
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self._speech = 0.0
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@property
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def active(self) -> bool:
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"""是否已经进入一段候选语音。"""
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return self._speech > 0.0
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def feed(self, pcm16: np.ndarray, chunk_ms: float) -> str | None:
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"""喂一块音频，返回 start / end / discard / None。"""
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if pcm16.size == 0:
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return None
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rms = float(np.sqrt(np.mean(pcm16.astype(np.float32) ** 2)))
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event = None
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if rms >= self.threshold:
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if self._speech == 0.0:
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event = "start"
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self._speech += chunk_ms
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self._silence = 0.0
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else:
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if self._speech > 0:
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self._silence += chunk_ms
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if self._silence >= self.silence_ms:
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if self._speech >= self.min_speech_ms:
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event = "end"
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else:
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event = "discard"
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self.reset()
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return event
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# ---------------------------------------------------------------- 引擎处理器
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def extract_user_text(messages: list) -> str:
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"""从 RTVI/OpenAI 形式的消息列表中取最后一条用户文本。"""
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for message in reversed(messages):
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if not isinstance(message, dict) or message.get("role") != "user":
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continue
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content = message.get("content", "")
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if isinstance(content, str):
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return content.strip()
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if isinstance(content, list):
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parts = [
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item.get("text", "")
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for item in content
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if isinstance(item, dict) and isinstance(item.get("text"), str)
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]
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return " ".join(part.strip() for part in parts if part.strip())
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return ""
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class EngineProcessor(FrameProcessor):
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"""把 ConversationEngine 包成一个 Pipecat 处理器。
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A 方案的核心假设在这里被检验：**一个处理器吞掉整条会话逻辑，
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Pipecat 只负责音频进出和帧路由。**
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"""
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def __init__(
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self,
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cfg: AppConfig,
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transport: SmallWebRTCTransport,
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rtvi: RTVIProcessor,
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):
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super().__init__()
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self.cfg = cfg
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self.transport = transport
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self.rtvi = rtvi
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self._vad = SimpleVAD()
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self._pre_roll: deque[bytes] = deque(maxlen=50) # 约 1 秒
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self._buf = bytearray() # 当前话语的 PCM16
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self._turn: asyncio.Task | None = None
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self._turn_kind: str | None = None
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self._engine: ConversationEngine | None = None
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self._engine_lock = asyncio.Lock()
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self._memory: SQLiteMemoryStore | None = None
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self._llm: FallbackProvider | None = None
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self._session = ""
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self._bot_speaking = False
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self._skip_tts = False
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# 计时：用于拿真实延迟数字
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self._t_speech_end = 0.0
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self._t_first_audio = 0.0
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# ------------------------------------------------------------ 生命周期
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async def _ensure_engine(self, *, require_stt: bool = False) -> ConversationEngine:
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async with self._engine_lock:
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if self._engine is None:
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self._memory = SQLiteMemoryStore(
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Path.home() / ".local/share/ai_assistant/memory.db")
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await self._memory.open()
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self._llm = FallbackProvider(create_provider(self.cfg.llm),
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create_provider(self.cfg.llm_fallback))
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# 文字首轮不应等待 STT 模型加载；第一次语音输入时再初始化。
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self._engine = ConversationEngine(
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self.cfg, self._llm, self._memory, ManualIdentityProvider(),
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create_tts(self.cfg.voice), stt=None)
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self._session = await self._engine.start_session(user_id="u_ylhan")
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if require_stt and self._engine.stt is None:
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self._engine.stt = StreamingSTT(self.cfg.voice)
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return self._engine
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async def setup(self, setup):
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await super().setup(setup)
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log.info("会话处理器已就绪")
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async def process_frame(self, frame, direction: FrameDirection):
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await super().process_frame(frame, direction)
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if isinstance(frame, StartFrame):
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await self.push_frame(frame, direction)
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elif isinstance(frame, InputAudioRawFrame):
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await self._on_audio(frame)
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elif isinstance(frame, LLMConfigureOutputFrame):
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# RTVI 在 send-text 前后用这个帧表达 audio_response 选项。
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self._skip_tts = frame.skip_tts
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await self.push_frame(frame, direction)
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elif isinstance(frame, LLMMessagesAppendFrame):
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text = extract_user_text(frame.messages)
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if text and frame.run_llm is not False:
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log.info("收到网页文字输入: %d 字", len(text))
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await self._cancel_turn()
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self._t_speech_end = time.perf_counter()
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self._t_first_audio = 0.0
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self._turn = self.create_task(
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self._run_turn(text=text, audio_response=not self._skip_tts),
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"conversation-text-turn",
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)
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self._turn_kind = "text"
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else:
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await self.push_frame(frame, direction)
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elif isinstance(frame, InterruptionFrame):
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# 客户端判定打断后发来的信号（或 Pipecat 内部产生）
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log.info("收到打断信号，取消当前生成")
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await self._cancel_turn()
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await self.push_frame(frame, direction)
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elif isinstance(frame, (EndFrame, CancelFrame)):
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await self._cancel_turn()
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await self.push_frame(frame, direction)
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else:
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await self.push_frame(frame, direction)
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# ------------------------------------------------------------ 音频处理
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async def _on_audio(self, frame: InputAudioRawFrame) -> None:
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pcm16 = np.frombuffer(frame.audio, dtype=np.int16)
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chunk_ms = len(pcm16) / IN_RATE * 1000.0
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# 头几帧打日志，用于确认音频真的到了处理器
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self._n_audio = getattr(self, "_n_audio", 0) + 1
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if self._n_audio <= 3:
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rms = float(np.sqrt(np.mean(pcm16.astype(np.float32) ** 2))) if pcm16.size else 0.0
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log.info("收到音频帧 #%d: %d 采样, %.0fms, RMS=%.0f",
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self._n_audio, pcm16.size, chunk_ms, rms)
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# 助手正在说话时收到的音频 = 用户插话的候选。
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# 这里只记日志，真正的打断判定在客户端（见方案 §5 的边界决定）。
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if self._bot_speaking and float(np.sqrt(np.mean(pcm16.astype(np.float32) ** 2))) > self._vad.threshold:
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log.info("检测到用户插话（助手正在播放）")
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was_active = self._vad.active
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self._pre_roll.append(frame.audio)
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event = self._vad.feed(pcm16, chunk_ms)
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if event == "start":
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# 带上说话前约 1 秒，避免 VAD 切掉首字。
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self._buf = bytearray().join(self._pre_roll)
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elif was_active:
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self._buf.extend(frame.audio)
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# 防止噪声或异常客户端让常驻进程无限积累音频。
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max_bytes = IN_RATE * 2 * 60
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if len(self._buf) > max_bytes:
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log.warning("单次话语超过 60 秒，强制截断处理")
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event = "end"
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self._vad.reset()
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if event == "discard":
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self._buf.clear()
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return
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if event == "end":
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audio = np.frombuffer(bytes(self._buf), dtype=np.int16).astype(np.float32) / 32768.0
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self._buf.clear()
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if len(audio) > IN_RATE * 0.3: # 太短的不处理
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# Playground 始终开着麦克风。文字轮次期间的环境声或扬声器回采
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# 不能抢占文字请求；真正的设备端打断将在 P2 由客户端判定。
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if (
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self._turn_kind == "text"
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and self._turn is not None
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and not self._turn.done()
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):
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log.info("文字轮次进行中，忽略并发麦克风话段")
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return
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self._t_speech_end = time.perf_counter()
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self._t_first_audio = 0.0
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await self._cancel_turn()
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self._turn = self.create_task(
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self._run_turn(audio=audio), "conversation-audio-turn")
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self._turn_kind = "audio"
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async def _cancel_turn(self) -> None:
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if self._turn and not self._turn.done():
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await self.cancel_task(self._turn, timeout=5.0)
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self._turn = None
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self._turn_kind = None
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# ------------------------------------------------------------ 跑一轮
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async def _run_turn(
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self,
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*,
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audio: np.ndarray | None = None,
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text: str | None = None,
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audio_response: bool = True,
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) -> None:
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llm_started = False
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reply_parts: list[str] = []
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try:
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engine = await self._ensure_engine(require_stt=audio is not None)
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if text is not None:
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# RTVIObserver 从标准上下文帧生成 user-llm-text，供网页
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# Conversation 区显示。实际会话历史仍由自有引擎管理。
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await self.push_frame(LLMContextFrame(
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context=LLMContext(messages=[
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{"role": "user", "content": text},
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]),
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))
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await self.rtvi.push_transport_message(
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RTVI.UserLLMTextMessage(data=RTVI.TextMessageData(text=text))
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)
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events = (
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engine.respond_to_text(text, session_id=self._session)
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if text is not None
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else engine.respond_to_audio(audio, session_id=self._session)
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)
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async for ev in events:
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if ev.kind == EventKind.TRANSCRIPT and ev.text:
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await self.push_frame(TranscriptionFrame(
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text=ev.text,
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user_id="u_ylhan",
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timestamp=datetime.now(UTC).isoformat(),
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finalized=True,
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))
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await self.rtvi.push_transport_message(
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RTVI.UserTranscriptionMessage(
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data=RTVI.UserTranscriptionMessageData(
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text=ev.text,
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user_id="u_ylhan",
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timestamp=datetime.now(UTC).isoformat(),
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final=True,
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)
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)
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)
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elif ev.kind == EventKind.RESPONSE_DELTA and ev.text:
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reply_parts.append(ev.text)
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if not llm_started:
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await self.push_frame(LLMFullResponseStartFrame())
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await self.rtvi.push_transport_message(RTVI.BotLLMStartedMessage())
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llm_started = True
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await self.push_frame(LLMTextFrame(text=ev.text))
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await self.rtvi.push_transport_message(
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RTVI.BotLLMTextMessage(data=RTVI.TextMessageData(text=ev.text))
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)
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elif (
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ev.kind == EventKind.SPEAK
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and audio_response
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and ev.audio is not None
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and len(ev.audio)
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):
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if self._t_first_audio == 0.0:
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self._t_first_audio = time.perf_counter()
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log.info("首音延迟（输入完成→首段音频）: %.0f ms",
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(self._t_first_audio - self._t_speech_end) * 1000)
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await self._push_audio(ev.audio, ev.sample_rate)
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elif ev.kind == EventKind.METRICS and ev.metrics:
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log.info("本轮指标: %s", ev.metrics.summary())
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except asyncio.CancelledError:
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log.info("本轮被取消")
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raise
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except Exception as e: # noqa: BLE001
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log.exception("跑轮失败: %s", e)
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finally:
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if llm_started:
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await self.push_frame(LLMFullResponseEndFrame())
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await self.rtvi.push_transport_message(RTVI.BotLLMStoppedMessage())
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reply = "".join(reply_parts).strip()
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if reply:
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# Playground 1.7 的 Conversation 视图主要消费完整转写/
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# bot-output；仅发送 token 级 bot-llm-text 不会稳定渲染。
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await self.rtvi.push_transport_message(
|
||
RTVI.BotTranscriptionMessage(
|
||
data=RTVI.TextMessageData(text=reply)
|
||
)
|
||
)
|
||
await self.rtvi.push_transport_message(
|
||
RTVI.BotOutputMessage(
|
||
data=RTVI.BotOutputMessageData(
|
||
text=reply,
|
||
aggregated_by="sentence",
|
||
spoken=audio_response,
|
||
)
|
||
)
|
||
)
|
||
|
||
async def _push_audio(self, samples: np.ndarray, sample_rate: int) -> None:
|
||
"""把 float32 音频推给传输层。必要时重采样到 OUT_RATE。"""
|
||
if sample_rate != OUT_RATE:
|
||
# 简单线性重采样。实验够用；正式实现应该用 soxr/sox（已装）。
|
||
n = int(len(samples) * OUT_RATE / sample_rate)
|
||
samples = np.interp(np.linspace(0, len(samples) - 1, n),
|
||
np.arange(len(samples)), samples).astype(np.float32)
|
||
pcm = (np.clip(samples, -1, 1) * 32767).astype(np.int16)
|
||
self._bot_speaking = True
|
||
# 按 20ms 一块推，模拟渐进播放
|
||
step = OUT_RATE // 50
|
||
for i in range(0, len(pcm), step):
|
||
chunk = pcm[i:i + step]
|
||
await self.push_frame(OutputAudioRawFrame(
|
||
audio=chunk.tobytes(), sample_rate=OUT_RATE, num_channels=1))
|
||
self._bot_speaking = False
|
||
|
||
async def cleanup(self):
|
||
await self._cancel_turn()
|
||
if self._engine is not None:
|
||
await self._engine.end_session()
|
||
if self._llm is not None:
|
||
await self._llm.aclose()
|
||
if self._memory is not None:
|
||
await self._memory.close()
|
||
self._engine = None
|
||
self._llm = None
|
||
self._memory = None
|
||
await super().cleanup()
|
||
|
||
|
||
# ---------------------------------------------------------------- HTTP 服务
|
||
|
||
app = FastAPI(title="AI Companion · Pipecat experimental server")
|
||
handler = SmallWebRTCRequestHandler()
|
||
_cfg: AppConfig | None = None
|
||
_tasks: list[PipelineWorker] = []
|
||
_bg_tasks: set[asyncio.Task] = set() # 持有后台 task 引用，避免被 GC
|
||
|
||
|
||
@app.post("/api/offer")
|
||
async def offer(request: SmallWebRTCRequest):
|
||
"""WebRTC 信令：客户端发 offer，我们回 answer。"""
|
||
|
||
async def on_connection(connection):
|
||
transport = SmallWebRTCTransport(
|
||
webrtc_connection=connection,
|
||
params=TransportParams(
|
||
audio_in_enabled=True,
|
||
audio_in_sample_rate=IN_RATE,
|
||
audio_in_channels=IN_CHANNELS,
|
||
audio_out_enabled=True,
|
||
audio_out_sample_rate=OUT_RATE,
|
||
audio_out_channels=1,
|
||
),
|
||
)
|
||
rtvi = RTVIProcessor()
|
||
engine_proc = EngineProcessor(_cfg, transport, rtvi)
|
||
pipeline = Pipeline([transport.input(), engine_proc, transport.output()])
|
||
# 注意：这个版本里 PipelineParams 没有 allow_interruptions——
|
||
# 打断在 1.10 已移到 pipecat/turns/ 框架。实验阶段我们不用它的
|
||
# 轮次框架（方案 §5 的边界决定是"打断由客户端判定"），
|
||
# 所以这里只开指标。
|
||
# 注意：PipelineWorker 的音频参数传构造函数；run() 另收
|
||
# WorkerParams（主要提供 task_manager）。漏传 run 参数会报：
|
||
# "PipelineWorker.run() missing 1 required positional argument: 'params'"
|
||
# 而这个异常发生在后台任务里，会被静默吞掉——
|
||
# 现象是"连接正常但什么反应都没有"，而且日志里一条错误都看不到。
|
||
# 所以下面的 _run_task 里必须 log.exception，不能让异常无声消失。
|
||
# 保留 RTVI：网页 prebuilt 客户端用它显示状态、转写和指标；
|
||
# N5095 的精简客户端可以忽略不认识的 data-channel 消息。
|
||
task = PipelineWorker(
|
||
pipeline,
|
||
params=PIPELINE_PARAMS,
|
||
enable_rtvi=True,
|
||
rtvi_processor=rtvi,
|
||
# EngineProcessor 显式发送这些事件，避免自有异步引擎产生的帧在
|
||
# observer 来源判定中丢失或重复。
|
||
rtvi_observer_params=RTVIObserverParams(
|
||
bot_llm_enabled=False,
|
||
user_llm_enabled=False,
|
||
user_transcription_enabled=False,
|
||
),
|
||
idle_timeout_secs=None,
|
||
)
|
||
_tasks.append(task)
|
||
log.info("新连接已建立，管线启动")
|
||
|
||
@task.event_handler("on_pipeline_started")
|
||
async def _on_pipeline_started(worker, frame):
|
||
log.info("管线已就绪")
|
||
|
||
@task.event_handler("on_pipeline_timeout")
|
||
async def _on_pipeline_timeout(worker, frame):
|
||
log.error("管线等待 %s 超时", type(frame).__name__)
|
||
|
||
@task.event_handler("on_pipeline_error")
|
||
async def _on_pipeline_error(worker, frame):
|
||
log.error("管线错误: %s", frame)
|
||
|
||
@transport.event_handler("on_client_connected")
|
||
async def _on_conn(t, client):
|
||
# PipelineWorker.run() 会自动发送且只应发送一次 StartFrame。
|
||
log.info("客户端已连接")
|
||
|
||
@transport.event_handler("on_client_disconnected")
|
||
async def _on_disc(t, client):
|
||
log.info("客户端断开")
|
||
await task.cancel()
|
||
|
||
# **必须作为后台任务跑，不能 await。**
|
||
# 这个回调是被 handle_web_request 调用的，而后者又在 HTTP 请求里被 await。
|
||
# 如果这里 `await task.run()`，就会在信令请求里阻塞整个会话的生命周期，
|
||
# 管线初始化的时序会歪掉——实测表现为 setup()/connect() 不被调用，
|
||
# 于是 on_client_connected 永远不触发、StartFrame 不发、一帧音频都收不到。
|
||
# 官方 runner 用的是 background_tasks.add_task(bot, args)，同样的道理。
|
||
async def _run_task():
|
||
log.info("PipelineWorker 开始运行")
|
||
try:
|
||
# TaskManager 必须在这里建——模块级没有运行中的事件循环
|
||
await task.run(WorkerParams(task_manager=TaskManager()))
|
||
except Exception as e: # noqa: BLE001
|
||
log.exception("PipelineWorker 异常退出: %s", e)
|
||
log.info("PipelineWorker 已结束")
|
||
|
||
bg = asyncio.create_task(_run_task())
|
||
_bg_tasks.add(bg)
|
||
bg.add_done_callback(_bg_tasks.discard)
|
||
|
||
answer = await handler.handle_web_request(request, on_connection)
|
||
if answer is None:
|
||
return {"status": "ok"}
|
||
return answer
|
||
|
||
|
||
@app.post("/start")
|
||
async def start_web_client():
|
||
"""给 Pipecat Playground 返回本服务的 WebRTC 信令入口。
|
||
|
||
prebuilt 网页客户端会先 POST /start，再用响应里的 endpoint 发 SDP offer；
|
||
自写 aiortc/N5095 客户端则可以继续直接调用 /api/offer。
|
||
"""
|
||
return {
|
||
"webrtcRequestParams": {
|
||
"endpoint": "/api/offer",
|
||
},
|
||
}
|
||
|
||
|
||
@app.get("/health")
|
||
async def health():
|
||
return {"status": "ok", "connections": len([t for t in _tasks if not t.has_finished()])}
|
||
|
||
|
||
@app.get("/")
|
||
async def root():
|
||
return RedirectResponse("/client/")
|
||
|
||
|
||
def mount_client() -> bool:
|
||
"""挂载 prebuilt 网页客户端（用于测试）。"""
|
||
try:
|
||
from pipecat_ai_small_webrtc_prebuilt.frontend import SmallWebRTCPrebuiltUI
|
||
app.mount("/client", SmallWebRTCPrebuiltUI, name="client")
|
||
return True
|
||
except Exception as e: # noqa: BLE001
|
||
log.warning("prebuilt 客户端不可用（不影响服务端）: %s", e)
|
||
return False
|
||
|
||
|
||
def check_runtime_data() -> None:
|
||
"""启动前验证运行时数据，避免首个会话里偷偷联网下载。"""
|
||
import nltk
|
||
|
||
try:
|
||
nltk.data.find("tokenizers/punkt_tab")
|
||
except LookupError as e:
|
||
raise RuntimeError(
|
||
"缺少 NLTK punkt_tab。请在部署阶段运行 "
|
||
"`python -m nltk.downloader punkt_tab`，并设置 NLTK_DATA。"
|
||
) from e
|
||
|
||
|
||
def main() -> int:
|
||
global _cfg
|
||
p = argparse.ArgumentParser()
|
||
p.add_argument("--host", default="0.0.0.0")
|
||
p.add_argument("--port", type=int, default=7860)
|
||
a = p.parse_args()
|
||
|
||
check_runtime_data()
|
||
_cfg = AppConfig.load()
|
||
log.info("LLM=%s TTS=%s/%s STT=%d线程",
|
||
_cfg.llm.model, _cfg.voice.tts_backend, _cfg.voice.tts_speaker,
|
||
_cfg.voice.stt_threads)
|
||
mounted = mount_client()
|
||
log.info("网页客户端: %s", "已挂载 /client/" if mounted else "未挂载")
|
||
|
||
import uvicorn
|
||
uvicorn.run(app, host=a.host, port=a.port, log_level="warning")
|
||
return 0
|
||
|
||
|
||
if __name__ == "__main__":
|
||
sys.exit(main())
|