wehub-resource-sync
750 lines
25 KiB
Python
750 lines
25 KiB
Python
import asyncio
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import json
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import websockets
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import time
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import numpy as np
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import argparse
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import ssl
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import os
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import wave
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import functools
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from concurrent.futures import ThreadPoolExecutor
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from scipy.spatial.distance import cosine
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import torch # 保留不影响
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def to_python(obj):
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"""递归地把 numpy / torch 等类型转成纯 Python，可 JSON 序列化。"""
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try:
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import numpy as np # noqa
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import torch # noqa
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except Exception:
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np = None
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torch = None
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if np is not None and isinstance(obj, np.generic):
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return obj.item()
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if np is not None and isinstance(obj, np.ndarray):
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return obj.tolist()
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if torch is not None and isinstance(obj, torch.Tensor):
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return obj.cpu().tolist()
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if isinstance(obj, dict):
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return {k: to_python(v) for k, v in obj.items()}
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if isinstance(obj, (list, tuple)):
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return [to_python(v) for v in obj]
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return obj
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parser = argparse.ArgumentParser()
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parser.add_argument("--host", type=str, default="0.0.0.0", required=False, help="host ip")
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parser.add_argument("--port", type=int, default=10095, required=False, help="grpc server port")
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parser.add_argument(
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"--asr_model",
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type=str,
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default="iic/speech_paraformer-large-contextual_asr_nat-zh-cn-16k-common-vocab8404",
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help="model from modelscope",
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)
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parser.add_argument("--asr_model_revision", type=str, default="v2.0.4", help="")
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parser.add_argument(
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"--asr_model_online",
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type=str,
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default="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online",
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help="model from modelscope",
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)
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parser.add_argument("--asr_model_online_revision", type=str, default="v2.0.4", help="")
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parser.add_argument(
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"--vad_model",
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type=str,
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default="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
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help="model from modelscope",
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)
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parser.add_argument("--vad_model_revision", type=str, default="v2.0.4", help="")
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parser.add_argument(
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"--punc_model",
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type=str,
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default="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
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help="model from modelscope",
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)
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parser.add_argument("--punc_model_revision", type=str, default="v2.0.4", help="")
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parser.add_argument("--ngpu", type=int, default=1, help="0 for cpu, 1 for gpu")
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parser.add_argument("--device", type=str, default="cuda", help="cuda, cpu")
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parser.add_argument("--ncpu", type=int, default=4, help="cpu cores")
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parser.add_argument(
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"--certfile",
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type=str,
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default="../../ssl_key/server.crt",
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required=False,
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help="certfile for ssl",
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)
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parser.add_argument(
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"--keyfile",
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type=str,
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default="../../ssl_key/server.key",
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required=False,
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help="keyfile for ssl",
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)
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# ====== 保存 2pass 离线阶段送入 ASR 的音频片段（排查 VAD 切分）======
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parser.add_argument(
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"--save_offline_segments",
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action="store_true",
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help="Save each offline (2pass) audio segment sent to offline ASR as wav for debugging VAD split.",
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)
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parser.add_argument(
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"--save_offline_segments_dir",
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type=str,
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default="./offline_segments",
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help="Directory to save offline wav segments when --save_offline_segments is enabled.",
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)
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# ====== 并发控制：核心新增 ======
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parser.add_argument(
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"--worker_threads",
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type=int,
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default=max(4, (os.cpu_count() or 4)),
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help="ThreadPoolExecutor max_workers. Used to offload blocking inference so event loop won't be blocked.",
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)
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parser.add_argument("--concurrent_vad", type=int, default=4, help="Max concurrent VAD generate() calls.")
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parser.add_argument("--concurrent_asr_online", type=int, default=4, help="Max concurrent streaming ASR generate() calls.")
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parser.add_argument("--concurrent_asr_offline", type=int, default=2, help="Max concurrent offline ASR generate() calls.")
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parser.add_argument("--concurrent_punc", type=int, default=1, help="Max concurrent punctuation generate() calls.")
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parser.add_argument("--concurrent_sv", type=int, default=1, help="Max concurrent speaker verification generate() calls.")
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parser.add_argument(
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"--speaker_db_reload_sec",
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type=int,
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default=5,
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help="Reload speaker_db.json at most once every N seconds (avoid frequent disk IO).",
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)
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args = parser.parse_args()
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websocket_users = set()
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SPEAKER_DB_PATH = os.path.join(os.path.dirname(__file__), "speaker_db.json")
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def _ensure_dir(p: str):
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try:
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os.makedirs(p, exist_ok=True)
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except Exception:
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pass
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def _pcm_duration_ms(pcm_bytes: bytes, fs: int, ch: int = 1, sampwidth: int = 2) -> int:
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"""根据 fs/ch/sampwidth 计算 PCM 时长，避免写死 16k -> 32 bytes/ms。"""
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if not pcm_bytes:
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return 0
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bytes_per_ms = (fs * ch * sampwidth) / 1000.0
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if bytes_per_ms <= 0:
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return 0
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return int(len(pcm_bytes) / bytes_per_ms)
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def _safe_int(v, default):
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try:
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return int(v)
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except Exception:
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return default
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# ========= speaker db：加缓存，避免每段都读盘 =========
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_SPEAKER_DB_CACHE = {}
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_SPEAKER_DB_CACHE_TS = 0.0
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def _load_speaker_db_sync():
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if not os.path.exists(SPEAKER_DB_PATH):
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return {}
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try:
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with open(SPEAKER_DB_PATH, "r", encoding="utf-8") as f:
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data = json.load(f)
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return data if isinstance(data, dict) else {}
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except Exception:
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return {}
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def get_speaker_db_cached(now_ts: float, reload_sec: int):
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global _SPEAKER_DB_CACHE, _SPEAKER_DB_CACHE_TS
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if (now_ts - _SPEAKER_DB_CACHE_TS) >= max(1, int(reload_sec)):
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_SPEAKER_DB_CACHE = _load_speaker_db_sync()
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_SPEAKER_DB_CACHE_TS = now_ts
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return _SPEAKER_DB_CACHE or {}
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def _save_wav_sync(out_path: str, audio_bytes: bytes, fs: int, ch: int, sampwidth: int):
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with wave.open(out_path, "wb") as wf:
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wf.setnchannels(ch)
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wf.setsampwidth(sampwidth)
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wf.setframerate(fs)
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wf.writeframes(audio_bytes)
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def save_offline_wav_segment_sync(websocket, audio_bytes: bytes, reason: str = "offline"):
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"""
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保存离线阶段送入 ASR 的音频片段，方便人工试听排查 VAD 切分是否正确。
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约定：audio_bytes 为 单声道 PCM16 little-endian（默认 16k）。
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（注意：这是同步函数，外层会放线程池执行）
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"""
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if not getattr(websocket, "save_offline_segments", False):
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return
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if "2pass" not in (getattr(websocket, "mode", "") or ""):
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return
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if not audio_bytes:
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return
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fs = int(getattr(websocket, "audio_fs", 16000) or 16000)
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ch = 1
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sampwidth = 2 # int16
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# int16 对齐
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if len(audio_bytes) % 2 == 1:
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audio_bytes = audio_bytes[:-1]
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if not audio_bytes:
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return
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seg_idx = int(getattr(websocket, "offline_seg_idx", 0))
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websocket.offline_seg_idx = seg_idx + 1
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duration_ms = _pcm_duration_ms(audio_bytes, fs=fs, ch=ch, sampwidth=sampwidth)
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base_dir = getattr(websocket, "offline_save_dir", args.save_offline_segments_dir)
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_ensure_dir(base_dir)
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wav_name = (getattr(websocket, "wav_name", "microphone") or "microphone").replace("/", "_")
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ts = int(time.time() * 1000)
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fname = f"{wav_name}_{ts}_seg{seg_idx:04d}_{reason}_{duration_ms}ms.wav"
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out_path = os.path.join(base_dir, fname)
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try:
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_save_wav_sync(out_path, audio_bytes, fs=fs, ch=ch, sampwidth=sampwidth)
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print(f"[SAVE_OFFLINE_SEG] {out_path} ({duration_ms} ms, {len(audio_bytes)} bytes)")
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except Exception as e:
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print(f"[SAVE_OFFLINE_SEG] failed: {e}")
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print("model loading")
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from funasr import AutoModel # noqa
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# ====== 离线 ASR ======
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model_asr = AutoModel(
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model="paraformer-zh",
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model_revision="v2.0.4",
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ngpu=args.ngpu,
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ncpu=args.ncpu,
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device=args.device,
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disable_pbar=True,
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disable_log=True,
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)
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# streaming asr
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model_asr_streaming = AutoModel(
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model=args.asr_model_online,
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model_revision=args.asr_model_online_revision,
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ngpu=args.ngpu,
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ncpu=args.ncpu,
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device=args.device,
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disable_pbar=True,
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disable_log=True,
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)
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# vad
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model_vad = AutoModel(
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model=args.vad_model,
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model_revision=args.vad_model_revision,
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ngpu=args.ngpu,
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ncpu=args.ncpu,
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device=args.device,
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disable_pbar=True,
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disable_log=True,
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)
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# punc
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if args.punc_model != "":
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model_punc = AutoModel(
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model=args.punc_model,
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model_revision=args.punc_model_revision,
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ngpu=args.ngpu,
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ncpu=args.ncpu,
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device=args.device,
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disable_pbar=True,
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disable_log=True,
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)
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else:
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model_punc = None
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# sv
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model_sv = AutoModel(
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model="iic/speech_campplus_sv_zh-cn_16k-common",
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ngpu=args.ngpu,
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device=args.device,
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disable_pbar=True,
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disable_log=True,
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)
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print("model loaded! (now supports multi-client with non-blocking inference)")
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# ====== 线程池 + 并发阈值（核心）======
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EXECUTOR = ThreadPoolExecutor(max_workers=int(args.worker_threads))
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SEM_VAD = asyncio.Semaphore(max(1, int(args.concurrent_vad)))
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SEM_ASR_ONLINE = asyncio.Semaphore(max(1, int(args.concurrent_asr_online)))
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SEM_ASR_OFFLINE = asyncio.Semaphore(max(1, int(args.concurrent_asr_offline)))
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SEM_PUNC = asyncio.Semaphore(max(1, int(args.concurrent_punc)))
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SEM_SV = asyncio.Semaphore(max(1, int(args.concurrent_sv)))
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SEM_WAV = asyncio.Semaphore(max(1, 4)) # 保存 wav 一般不需要太大
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async def run_blocking(fn, *a, sem: asyncio.Semaphore | None = None, **kw):
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"""
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把阻塞函数丢线程池执行，避免卡 event loop。
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sem 用于限流（避免 GPU / 模型被打爆）。
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"""
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loop = asyncio.get_running_loop()
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call = functools.partial(fn, *a, **kw)
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if sem is None:
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return await loop.run_in_executor(EXECUTOR, call)
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async with sem:
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return await loop.run_in_executor(EXECUTOR, call)
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def _generate_sync(model, audio_or_text, status_dict):
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# 注意：status_dict 里包含 cache，会被 generate 更新
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return model.generate(input=audio_or_text, **status_dict)
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async def ws_reset(websocket):
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print("ws reset now, total num is ", len(websocket_users))
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websocket.status_dict_asr_online["cache"] = {}
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websocket.status_dict_asr_online["is_final"] = True
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websocket.status_dict_vad["cache"] = {}
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websocket.status_dict_vad["is_final"] = True
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websocket.status_dict_punc["cache"] = {}
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await websocket.close()
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async def clear_websocket():
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for websocket in list(websocket_users):
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await ws_reset(websocket)
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websocket_users.clear()
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async def ws_serve(websocket, path=None):
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# websockets 新版本不会传 path，这里做兼容
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if path is None:
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path = getattr(websocket, "path", None)
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frames = []
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frames_asr = []
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frames_asr_online = []
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global websocket_users
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websocket_users.add(websocket)
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websocket.status_dict_asr = {} # hotword 等
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websocket.status_dict_asr_online = {"cache": {}, "is_final": False}
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websocket.status_dict_vad = {"cache": {}, "is_final": False}
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websocket.status_dict_punc = {"cache": {}}
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websocket.chunk_interval = 10
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websocket.vad_pre_idx = 0
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speech_start = False
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speech_end_i = -1
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websocket.wav_name = "microphone"
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websocket.mode = "2pass"
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websocket.is_speaking = True # ✅ 默认初始化，避免 AttributeError
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# 保存离线片段
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websocket.audio_fs = 16000
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websocket.offline_seg_idx = 0
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websocket.save_offline_segments = bool(args.save_offline_segments)
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websocket.offline_save_dir = args.save_offline_segments_dir
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if websocket.save_offline_segments:
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_ensure_dir(websocket.offline_save_dir)
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print(f"[SAVE_OFFLINE_SEG] enabled, dir={websocket.offline_save_dir}")
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print("new user connected", flush=True)
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try:
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async for message in websocket:
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# ========== 1) 先处理“文本配置消息” ==========
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if isinstance(message, str):
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try:
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messagejson = json.loads(message)
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except Exception as e:
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print("bad json message:", e, message[:200])
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continue
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print("=============messagejson============", messagejson)
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if "is_speaking" in messagejson:
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websocket.is_speaking = bool(messagejson["is_speaking"])
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websocket.status_dict_asr_online["is_final"] = (not websocket.is_speaking)
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if "chunk_interval" in messagejson:
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websocket.chunk_interval = _safe_int(
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messagejson["chunk_interval"], websocket.chunk_interval
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)
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if "wav_name" in messagejson:
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websocket.wav_name = messagejson.get("wav_name") or websocket.wav_name
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if "chunk_size" in messagejson:
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chunk_size = messagejson["chunk_size"]
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if isinstance(chunk_size, str):
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chunk_size = [x.strip() for x in chunk_size.split(",") if x.strip()]
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websocket.status_dict_asr_online["chunk_size"] = [int(x) for x in chunk_size]
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if "encoder_chunk_look_back" in messagejson:
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websocket.status_dict_asr_online["encoder_chunk_look_back"] = messagejson[
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"encoder_chunk_look_back"
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]
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if "decoder_chunk_look_back" in messagejson:
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websocket.status_dict_asr_online["decoder_chunk_look_back"] = messagejson[
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"decoder_chunk_look_back"
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]
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if "hotwords" in messagejson:
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hotword_data = messagejson["hotwords"]
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websocket.status_dict_asr["hotword"] = hotword_data
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websocket.status_dict_asr_online["hotword"] = hotword_data
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print(f"热词已更新: {hotword_data}")
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if "mode" in messagejson:
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websocket.mode = messagejson["mode"] or websocket.mode
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if "audio_fs" in messagejson:
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websocket.audio_fs = _safe_int(messagejson["audio_fs"], 16000)
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continue
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# ========== 2) 处理“二进制音频消息” ==========
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if "chunk_size" not in websocket.status_dict_asr_online:
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print("[WARN] chunk_size not set yet, skip audio frame (send config first).")
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continue
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try:
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websocket.status_dict_vad["chunk_size"] = int(
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websocket.status_dict_asr_online["chunk_size"][1] * 60 / websocket.chunk_interval
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)
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except Exception as e:
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print("[WARN] set vad chunk_size failed:", e)
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continue
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pcm = message
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frames.append(pcm)
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duration_ms = _pcm_duration_ms(pcm, fs=websocket.audio_fs, ch=1, sampwidth=2)
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websocket.vad_pre_idx += duration_ms
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# online asr
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frames_asr_online.append(pcm)
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websocket.status_dict_asr_online["is_final"] = (speech_end_i != -1)
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if (len(frames_asr_online) % websocket.chunk_interval == 0) or websocket.status_dict_asr_online["is_final"]:
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if websocket.mode in ("2pass", "online"):
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audio_in = b"".join(frames_asr_online)
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try:
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await async_asr_online(websocket, audio_in)
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except Exception:
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print(f"error in asr streaming, {websocket.status_dict_asr_online}")
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frames_asr_online = []
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if speech_start:
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frames_asr.append(pcm)
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# vad online
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try:
|
||
speech_start_i, speech_end_i = await async_vad(websocket, pcm)
|
||
except Exception as e:
|
||
print("error in vad:", e)
|
||
speech_start_i, speech_end_i = -1, -1
|
||
|
||
if speech_start_i != -1:
|
||
speech_start = True
|
||
if duration_ms > 0:
|
||
beg_bias = (websocket.vad_pre_idx - speech_start_i) // duration_ms
|
||
else:
|
||
beg_bias = 0
|
||
frames_pre = frames[-beg_bias:] if beg_bias > 0 else []
|
||
frames_asr = []
|
||
frames_asr.extend(frames_pre)
|
||
|
||
# ========== 3) 2pass：离线阶段触发点 ==========
|
||
if (speech_end_i != -1) or (not websocket.is_speaking):
|
||
if websocket.mode in ("2pass", "offline"):
|
||
audio_in = b"".join(frames_asr)
|
||
reason = "vad_end" if speech_end_i != -1 else "not_speaking"
|
||
|
||
# 保存 wav：放线程池，避免磁盘 IO 卡 loop
|
||
if websocket.save_offline_segments and audio_in:
|
||
try:
|
||
await run_blocking(
|
||
save_offline_wav_segment_sync,
|
||
websocket,
|
||
audio_in,
|
||
reason,
|
||
sem=SEM_WAV,
|
||
)
|
||
except Exception as e:
|
||
print("[SAVE_OFFLINE_SEG] async failed:", e)
|
||
|
||
try:
|
||
await async_asr(websocket, audio_in)
|
||
except Exception as e:
|
||
print("error in asr offline:", e)
|
||
|
||
frames_asr = []
|
||
speech_start = False
|
||
frames_asr_online = []
|
||
websocket.status_dict_asr_online["cache"] = {}
|
||
|
||
if not websocket.is_speaking:
|
||
websocket.vad_pre_idx = 0
|
||
frames = []
|
||
websocket.status_dict_vad["cache"] = {}
|
||
speech_end_i = -1
|
||
else:
|
||
frames = frames[-20:]
|
||
|
||
except websockets.ConnectionClosed:
|
||
print("ConnectionClosed...", websocket_users, flush=True)
|
||
await ws_reset(websocket)
|
||
if websocket in websocket_users:
|
||
websocket_users.remove(websocket)
|
||
except websockets.InvalidState:
|
||
print("InvalidState...")
|
||
except Exception as e:
|
||
print("Exception:", e)
|
||
try:
|
||
await ws_reset(websocket)
|
||
except Exception:
|
||
pass
|
||
if websocket in websocket_users:
|
||
websocket_users.remove(websocket)
|
||
|
||
|
||
# ===================== 推理：全部改为“线程池 + 限流” =====================
|
||
|
||
async def async_vad(websocket, audio_in: bytes):
|
||
# model_vad.generate 是阻塞的，必须 offload
|
||
out = await run_blocking(_generate_sync, model_vad, audio_in, websocket.status_dict_vad, sem=SEM_VAD)
|
||
segments_result = out[0].get("value", [])
|
||
|
||
speech_start = -1
|
||
speech_end = -1
|
||
|
||
if len(segments_result) == 0 or len(segments_result) > 1:
|
||
return speech_start, speech_end
|
||
if segments_result[0][0] != -1:
|
||
speech_start = segments_result[0][0]
|
||
if segments_result[0][1] != -1:
|
||
speech_end = segments_result[0][1]
|
||
return speech_start, speech_end
|
||
|
||
|
||
def _sv_and_match_sync(audio_in: bytes, reload_sec: int):
|
||
"""
|
||
同步执行：SV embedding + speaker_db 匹配
|
||
返回 (spk_name, best_score)
|
||
"""
|
||
spk_name = "unknown"
|
||
best_score = 0.0
|
||
|
||
sv_out = model_sv.generate(input=audio_in, embedding=True)[0]
|
||
embedding = sv_out["spk_embedding"][0].cpu().numpy()
|
||
|
||
now_ts = time.time()
|
||
local_speaker_db = get_speaker_db_cached(now_ts, reload_sec=reload_sec)
|
||
if local_speaker_db:
|
||
for name, ref_embedding in local_speaker_db.items():
|
||
if ref_embedding is None:
|
||
continue
|
||
arr = np.array(ref_embedding, dtype=np.float32)
|
||
similarity = 1.0 - cosine(embedding, arr)
|
||
print("sv similarity with {}: {}".format(name, similarity))
|
||
if similarity > best_score and similarity > 0.2:
|
||
best_score = similarity
|
||
spk_name = name
|
||
|
||
return spk_name, float(best_score)
|
||
|
||
|
||
async def async_asr(websocket, audio_in: bytes):
|
||
mode = "2pass-offline" if "2pass" in (websocket.mode or "") else websocket.mode
|
||
|
||
if len(audio_in) <= 0:
|
||
message = {
|
||
"mode": mode,
|
||
"text": "",
|
||
"wav_name": websocket.wav_name,
|
||
"is_final": True,
|
||
}
|
||
await websocket.send(json.dumps(message, ensure_ascii=False))
|
||
return
|
||
|
||
# 1) ASR（阻塞，线程池执行）
|
||
rec_result_list = await run_blocking(
|
||
_generate_sync,
|
||
model_asr,
|
||
audio_in,
|
||
websocket.status_dict_asr,
|
||
sem=SEM_ASR_OFFLINE,
|
||
)
|
||
rec_result = rec_result_list[0]
|
||
|
||
print("offline_asr, raw:", rec_result)
|
||
print("offline_asr, keys:", rec_result.keys())
|
||
|
||
text = rec_result.get("text", "")
|
||
timestamp = rec_result.get("timestamp", None)
|
||
sentence_info = rec_result.get("sentence_info", None)
|
||
|
||
# 2) 声纹识别（阻塞，线程池执行）
|
||
spk_name = "unknown"
|
||
best_score = 0.0
|
||
try:
|
||
spk_name, best_score = await run_blocking(
|
||
_sv_and_match_sync,
|
||
audio_in,
|
||
int(args.speaker_db_reload_sec),
|
||
sem=SEM_SV,
|
||
)
|
||
except Exception as e:
|
||
print(f"声纹识别失败: {e}")
|
||
|
||
# 3) 标点（阻塞，线程池执行）
|
||
punc_array = None
|
||
if model_punc is not None and len(text) > 0:
|
||
try:
|
||
# punc 只对文本处理
|
||
punc_out = await run_blocking(
|
||
_generate_sync,
|
||
model_punc,
|
||
text,
|
||
websocket.status_dict_punc,
|
||
sem=SEM_PUNC,
|
||
)
|
||
punc_result = punc_out[0]
|
||
print("offline, after punc", punc_result)
|
||
|
||
if "text" in punc_result and punc_result["text"]:
|
||
text = punc_result["text"]
|
||
if "punc_array" in punc_result:
|
||
punc_array = punc_result["punc_array"]
|
||
except Exception as e:
|
||
print("punc failed:", e)
|
||
|
||
# 4) 构造最终 message
|
||
if len(text) > 0:
|
||
print("======offline final text:", text)
|
||
message = {
|
||
"mode": mode,
|
||
"spk_name": spk_name,
|
||
"spk_score": float(best_score),
|
||
"text": text,
|
||
"wav_name": websocket.wav_name,
|
||
"is_final": True,
|
||
}
|
||
if timestamp is not None:
|
||
message["timestamp"] = to_python(timestamp)
|
||
if sentence_info is not None:
|
||
message["sentence_info"] = to_python(sentence_info)
|
||
if punc_array is not None:
|
||
message["punc_array"] = to_python(punc_array)
|
||
|
||
try:
|
||
await websocket.send(json.dumps(message, ensure_ascii=False))
|
||
except Exception as e:
|
||
print("send json failed:", e)
|
||
print("message types:", {k: type(v) for k, v in message.items()})
|
||
else:
|
||
message = {
|
||
"mode": mode,
|
||
"spk_name": spk_name,
|
||
"spk_score": float(best_score),
|
||
"text": "",
|
||
"wav_name": websocket.wav_name,
|
||
"is_final": True,
|
||
}
|
||
await websocket.send(json.dumps(message, ensure_ascii=False))
|
||
|
||
|
||
async def async_asr_online(websocket, audio_in: bytes):
|
||
if len(audio_in) <= 0:
|
||
return
|
||
|
||
# streaming generate 也是阻塞：线程池执行
|
||
rec_out = await run_blocking(
|
||
_generate_sync,
|
||
model_asr_streaming,
|
||
audio_in,
|
||
websocket.status_dict_asr_online,
|
||
sem=SEM_ASR_ONLINE,
|
||
)
|
||
rec_result = rec_out[0]
|
||
print("online, ", rec_result)
|
||
|
||
# 2pass：online 只要 partial，不发 final（final 交给 offline）
|
||
if websocket.mode == "2pass" and websocket.status_dict_asr_online.get("is_final", False):
|
||
return
|
||
|
||
if rec_result.get("text"):
|
||
mode = "2pass-online" if "2pass" in (websocket.mode or "") else websocket.mode
|
||
message = {
|
||
"mode": mode,
|
||
"text": rec_result["text"],
|
||
"wav_name": websocket.wav_name,
|
||
"is_final": bool(
|
||
websocket.status_dict_asr_online.get("is_final", False) or (not websocket.is_speaking)
|
||
),
|
||
}
|
||
await websocket.send(json.dumps(message, ensure_ascii=False))
|
||
|
||
|
||
# ===================== 启动服务 =====================
|
||
|
||
async def main():
|
||
if len(args.certfile) > 0:
|
||
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER)
|
||
ssl_context.load_cert_chain(args.certfile, keyfile=args.keyfile)
|
||
server = await websockets.serve(
|
||
ws_serve,
|
||
args.host,
|
||
args.port,
|
||
subprotocols=["binary"],
|
||
ping_interval=None,
|
||
ssl=ssl_context,
|
||
)
|
||
else:
|
||
server = await websockets.serve(
|
||
ws_serve,
|
||
args.host,
|
||
args.port,
|
||
subprotocols=["binary"],
|
||
ping_interval=None,
|
||
)
|
||
|
||
print(f"WS server started at ws(s)://{args.host}:{args.port}")
|
||
await server.wait_closed()
|
||
|
||
|
||
if __name__ == "__main__":
|
||
try:
|
||
asyncio.run(main())
|
||
finally:
|
||
try:
|
||
EXECUTOR.shutdown(wait=False, cancel_futures=True)
|
||
except Exception:
|
||
pass
|