wehub-resource-sync
558 lines
20 KiB
Python
558 lines
20 KiB
Python
# -*- encoding: utf-8 -*-
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import os
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import time
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import websockets, ssl
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import asyncio
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import argparse
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import json
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import traceback
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from multiprocessing import Process
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import logging
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logging.basicConfig(level=logging.ERROR)
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"--host", type=str, default="localhost", required=False, help="host ip, localhost, 0.0.0.0"
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)
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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("--chunk_size", type=str, default="5, 10, 5", help="chunk")
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parser.add_argument("--encoder_chunk_look_back", type=int, default=4, help="chunk")
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parser.add_argument("--decoder_chunk_look_back", type=int, default=0, help="chunk")
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parser.add_argument("--chunk_interval", type=int, default=10, help="chunk")
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parser.add_argument(
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"--hotword",
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type=str,
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default="",
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help="hotword file path, one hotword perline (e.g.:阿里巴巴 20)",
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)
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parser.add_argument(
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"--audio_in",
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type=str,
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default=None,
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help="音频输入路径；不传则使用麦克风（需安装 PyAudio）",
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)
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parser.add_argument("--audio_fs", type=int, default=16000, help="audio_fs")
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# ✅ 修复语义：默认 False；传入参数则不 sleep（用于压测）
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parser.add_argument(
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"--send_without_sleep",
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action="store_true",
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default=False,
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help="若设置：发送音频不按实时节奏 sleep（用于压测）",
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)
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parser.add_argument("--thread_num", type=int, default=1, help="thread_num")
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parser.add_argument("--words_max_print", type=int, default=10000, help="chunk")
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parser.add_argument("--output_dir", type=str, default=None, help="output_dir")
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parser.add_argument("--ssl", type=int, default=1, help="1 for ssl connect, 0 for no ssl")
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parser.add_argument("--use_itn", type=int, default=1, help="1 for using itn, 0 for not itn")
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parser.add_argument("--mode", type=str, default="2pass", help="offline, online, 2pass")
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# ✅ 验收日志输出目录（每个 meeting 单独写，避免多进程抢文件）
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parser.add_argument("--log_dir", type=str, default="./asr_logs", help="验收日志输出目录")
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parser.add_argument("--log_flush_every", type=int, default=1, help="events.jsonl 每写N行flush一次（默认1更安全）")
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args = parser.parse_args()
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args.chunk_size = [int(x) for x in args.chunk_size.split(",")]
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print(args)
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from queue import Queue
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from datetime import datetime
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voices = Queue()
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offline_msg_done = False
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# === 延迟统计相关：对每个 wav_name 记录首包/末包发送时间 & 是否已经打印过延迟 ===
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latency_first_audio_time = {} # {wav_name: t_first_chunk_send}
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latency_last_audio_time = {} # {wav_name: t_last_chunk_send}
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latency_first_text_printed = {} # {wav_name: bool}
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def _iso(ts: float) -> str:
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return datetime.fromtimestamp(ts).strftime("%Y-%m-%d %H:%M:%S.%f")[:-3]
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class MeetingWriter:
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"""
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每个进程/meeting 单独写：
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- events.jsonl：收到的每条服务端消息（在线/离线/2pass）
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- meta.json：本次运行参数（方便复现）
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"""
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def __init__(self, log_dir: str, meeting_id: str, flush_every: int = 1):
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self.meeting_id = str(meeting_id)
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self.base = os.path.join(log_dir, f"meeting_{self.meeting_id}")
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os.makedirs(self.base, exist_ok=True)
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self.fp_events = open(os.path.join(self.base, "events.jsonl"), "a", encoding="utf-8")
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self.flush_every = max(1, int(flush_every))
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self._cnt = 0
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meta_path = os.path.join(self.base, "meta.json")
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if not os.path.exists(meta_path):
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with open(meta_path, "w", encoding="utf-8") as f:
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meta = {
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"created_at": _iso(time.time()),
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"meeting_id": self.meeting_id,
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"args": vars(args),
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}
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f.write(json.dumps(meta, ensure_ascii=False, indent=2))
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def write_event(self, obj: dict):
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self.fp_events.write(json.dumps(obj, ensure_ascii=False) + "\n")
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self._cnt += 1
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if self._cnt % self.flush_every == 0:
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self.fp_events.flush()
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def close(self):
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try:
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self.fp_events.flush()
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self.fp_events.close()
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except Exception:
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pass
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if args.output_dir is not None:
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if not os.path.exists(args.output_dir):
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os.makedirs(args.output_dir)
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async def record_microphone():
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"""从麦克风实时录音发送到服务端（一般单路测试使用）"""
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try:
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import pyaudio
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except ImportError as e:
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raise ImportError(
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"缺少 PyAudio，麦克风推流前请先运行 `pip install pyaudio`"
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) from e
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global voices
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FORMAT = pyaudio.paInt16
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CHANNELS = 1
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RATE = 16000
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chunk_size = 60 * args.chunk_size[1] / args.chunk_interval
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CHUNK = int(RATE / 1000 * chunk_size)
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p = pyaudio.PyAudio()
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stream = p.open(
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format=FORMAT, channels=CHANNELS, rate=RATE, input=True, frames_per_buffer=CHUNK
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)
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# hotwords
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fst_dict = {}
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hotword_msg = ""
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if args.hotword.strip() != "":
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if os.path.exists(args.hotword):
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f_scp = open(args.hotword, encoding="utf-8")
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hot_lines = f_scp.readlines()
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for line in hot_lines:
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words = line.strip().split(" ")
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if len(words) < 2:
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print("Please checkout format of hotwords")
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continue
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try:
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fst_dict[" ".join(words[:-1])] = int(words[-1])
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except ValueError:
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print("Please checkout format of hotwords")
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hotword_msg = json.dumps(fst_dict, ensure_ascii=False)
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else:
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hotword_msg = args.hotword
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use_itn = True
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if args.use_itn == 0:
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use_itn = False
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message = json.dumps(
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{
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"mode": args.mode,
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"chunk_size": args.chunk_size,
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"chunk_interval": args.chunk_interval,
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"encoder_chunk_look_back": args.encoder_chunk_look_back,
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"decoder_chunk_look_back": args.decoder_chunk_look_back,
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"wav_name": "microphone",
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"is_speaking": True,
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"hotwords": hotword_msg,
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"itn": use_itn,
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},
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ensure_ascii=False,
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)
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await websocket.send(message)
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while True:
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data = stream.read(CHUNK)
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await websocket.send(data)
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await asyncio.sleep(0.01)
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async def record_from_scp(chunk_begin, chunk_size):
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"""从 wav/scp 文件读取音频分片发送，用于压测和延迟测试"""
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global voices, latency_first_audio_time, latency_last_audio_time
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if args.audio_in.endswith(".scp"):
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f_scp = open(args.audio_in)
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wavs = f_scp.readlines()
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else:
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wavs = [args.audio_in]
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# hotwords
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hotword_msg = ""
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if args.hotword.strip() != "":
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if os.path.exists(args.hotword):
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with open(args.hotword, encoding="utf-8") as f_scp:
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hot_lines = f_scp.readlines()
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hot_list = []
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for line in hot_lines:
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words = line.strip().split()
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if not words:
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continue
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# Python AutoModel: 用逗号分隔多个热词
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hot_list.append(words[0])
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hotword_msg = ",".join(hot_list)
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else:
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hotword_msg = args.hotword
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print("hotword", hotword_msg)
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sample_rate = args.audio_fs
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wav_format = "pcm"
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use_itn = True
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if args.use_itn == 0:
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use_itn = False
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if chunk_size > 0:
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wavs = wavs[chunk_begin: chunk_begin + chunk_size]
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for wav in wavs:
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wav_splits = wav.strip().split()
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wav_name = wav_splits[0] if len(wav_splits) > 1 else "demo"
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wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
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if not len(wav_path.strip()) > 0:
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continue
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if wav_path.endswith(".pcm"):
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with open(wav_path, "rb") as f:
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audio_bytes = f.read()
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elif wav_path.endswith(".wav"):
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import wave
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with wave.open(wav_path, "rb") as wav_file:
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sample_rate = wav_file.getframerate()
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frames = wav_file.readframes(wav_file.getnframes())
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audio_bytes = bytes(frames)
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else:
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wav_format = "others"
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with open(wav_path, "rb") as f:
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audio_bytes = f.read()
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stride = int(60 * args.chunk_size[1] / args.chunk_interval / 1000 * sample_rate * 2)
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chunk_num = (len(audio_bytes) - 1) // stride + 1
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# send first control message
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message = json.dumps(
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{
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"mode": args.mode,
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"chunk_size": args.chunk_size,
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"chunk_interval": args.chunk_interval,
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"encoder_chunk_look_back": args.encoder_chunk_look_back,
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"decoder_chunk_look_back": args.decoder_chunk_look_back,
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"audio_fs": sample_rate,
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"wav_name": wav_name,
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"wav_format": wav_format,
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"is_speaking": True,
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"hotwords": hotword_msg,
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"itn": use_itn,
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},
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ensure_ascii=False,
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)
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await websocket.send(message)
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is_speaking = True
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# 初始化该 wav 的统计状态
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latency_first_audio_time[wav_name] = None
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latency_last_audio_time[wav_name] = None
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latency_first_text_printed[wav_name] = False
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for i in range(chunk_num):
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beg = i * stride
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data = audio_bytes[beg: beg + stride]
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now_ts = time.time()
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if latency_first_audio_time[wav_name] is None:
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latency_first_audio_time[wav_name] = now_ts
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latency_last_audio_time[wav_name] = now_ts
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await websocket.send(data)
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if i == chunk_num - 1:
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is_speaking = False
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await websocket.send(json.dumps({"is_speaking": is_speaking}, ensure_ascii=False))
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# ✅ sleep策略：默认按实时节奏；若开启 send_without_sleep 则几乎不 sleep（压测）
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if args.send_without_sleep:
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sleep_duration = 0.001
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else:
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sleep_duration = (
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0.001
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if args.mode == "offline"
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else 60 * args.chunk_size[1] / args.chunk_interval / 1000
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)
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await asyncio.sleep(sleep_duration)
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if not args.mode == "offline":
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await asyncio.sleep(2)
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if args.mode == "offline":
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global offline_msg_done
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while not offline_msg_done:
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await asyncio.sleep(1)
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await asyncio.sleep(10)
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await websocket.close()
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async def message(id, writer: MeetingWriter):
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"""接收服务端识别结果 + 打印实时文本 + 打印延迟 + 写验收日志(events.jsonl)"""
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import websockets
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global websocket, voices, offline_msg_done
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global latency_first_audio_time, latency_last_audio_time, latency_first_text_printed
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multi_mode = args.thread_num > 1 # 多路并发时，打印风格更简洁
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text_print = ""
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text_print_2pass_online = ""
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text_print_2pass_offline = ""
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if args.output_dir is not None:
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ibest_writer = open(
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os.path.join(args.output_dir, "text.{}".format(id)), "a", encoding="utf-8"
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)
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else:
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ibest_writer = None
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try:
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while True:
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meg = await websocket.recv()
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meg = json.loads(meg)
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wav_name = meg.get("wav_name", "demo")
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text = meg.get("text", "")
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mode = meg.get("mode", "")
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spk_name = meg.get("spk_name", "")
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spk_score = meg.get("spk_score", None)
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now_ts = time.time()
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# === 延迟统计：仅在首条 online/2pass-online 文本时计算并打印一次 ===
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latency_last_ms = None
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latency_first_ms = None
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if text and mode in ("online", "2pass-online"):
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if not latency_first_text_printed.get(wav_name, False):
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t_last = latency_last_audio_time.get(wav_name, None)
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t_first = latency_first_audio_time.get(wav_name, None)
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latency_last_ms = (now_ts - t_last) * 1000.0 if t_last is not None else None
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latency_first_ms = (now_ts - t_first) * 1000.0 if t_first is not None else None
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latency_first_text_printed[wav_name] = True
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if multi_mode:
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parts = [f"[MEETING {id}][LATENCY] wav={wav_name}, mode={mode}"]
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if latency_last_ms is not None:
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parts.append(f"from_last_chunk={latency_last_ms:.1f} ms")
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if latency_first_ms is not None:
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parts.append(f"from_first_chunk={latency_first_ms:.1f} ms")
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print(" ".join(parts))
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else:
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print(
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f"[LATENCY] wav={wav_name}, mode={mode}, "
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f"from_last_chunk={(latency_last_ms or 0):.1f} ms, "
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f"from_first_chunk={(latency_first_ms or 0):.1f} ms"
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)
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timestamp = meg.get("timestamp", "")
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offline_msg_done = meg.get("is_final", False)
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# ✅ 验收友好：每条消息落 events.jsonl（便于后处理）
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event = {
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"ts": _iso(now_ts),
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"recv_ts": now_ts,
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"meeting_id": str(id),
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"wav_name": wav_name,
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"mode": mode,
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"is_final": bool(meg.get("is_final", False)),
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"text": text,
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"spk_name": spk_name,
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"spk_score": spk_score,
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"latency_first_ms": latency_first_ms,
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"latency_last_ms": latency_last_ms,
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"server_timestamp": meg.get("timestamp", None),
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"sentence_info": meg.get("sentence_info", None),
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"punc_array": meg.get("punc_array", None),
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}
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if writer is not None:
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writer.write_event(event)
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# 保存到 output_dir（保留你原来的逻辑）
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if ibest_writer is not None and text:
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if timestamp != "":
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text_write_line = "{}\t{}\t{}\n".format(wav_name, text, timestamp)
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else:
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text_write_line = "{}\t{}\n".format(wav_name, text)
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ibest_writer.write(text_write_line)
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if "mode" not in meg:
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continue
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# ===== 多路并发输出风格：只打印精简行 =====
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if multi_mode:
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if mode in ("offline", "2pass-offline") and text:
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spk_name2 = meg.get("spk_name", "unknown")
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spk_score2 = meg.get("spk_score", 0.0)
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print(
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f"[MEETING {id}][FINAL][{wav_name}] "
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f"spk={spk_name2}({float(spk_score2):.3f}) text=\"{text}\""
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)
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if timestamp:
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print(f"[MEETING {id}][TIMESTAMP][{wav_name}] {timestamp}")
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continue
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# ===== 单路模式输出：保留原滚动体验 =====
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if meg["mode"] == "online":
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text_print += "{}".format(text)
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text_print = text_print[-args.words_max_print:]
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print("pid" + str(id) + ": " + text_print)
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elif meg["mode"] == "offline":
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if timestamp != "":
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text_print += "{} timestamp: {}".format(text, timestamp)
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else:
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text_print += "{}".format(text)
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spk_info = ""
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if spk_name:
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if spk_score is not None:
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spk_info = f" [spk={spk_name} score={float(spk_score):.3f}]"
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else:
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spk_info = f" [spk={spk_name}]"
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print("pid" + str(id) + ": " + wav_name + ": " + text_print + spk_info)
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offline_msg_done = True
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else:
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# 2pass 模式
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if meg["mode"] == "2pass-online":
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text_print_2pass_online += "{}".format(text)
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text_print = text_print_2pass_offline + text_print_2pass_online
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else:
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text_print_2pass_online = ""
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text_print = text_print_2pass_offline + "{}".format(text)
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text_print_2pass_offline += "{}".format(text)
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if spk_name:
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if spk_score is not None:
|
|
text_print += f" [spk={spk_name} score={float(spk_score):.3f}]"
|
|
else:
|
|
text_print += f" [spk={spk_name}]"
|
|
|
|
text_print = text_print[-args.words_max_print:]
|
|
print("pid" + str(id) + ": " + text_print)
|
|
|
|
except websockets.exceptions.ConnectionClosedOK:
|
|
print(f"[MEETING {id}] connection closed normally")
|
|
except Exception as e:
|
|
print(f"[MEETING {id}] Exception:", e)
|
|
finally:
|
|
try:
|
|
if ibest_writer is not None:
|
|
ibest_writer.flush()
|
|
ibest_writer.close()
|
|
except Exception:
|
|
pass
|
|
|
|
|
|
async def ws_client(id, chunk_begin, chunk_size):
|
|
if args.audio_in is None:
|
|
chunk_begin = 0
|
|
chunk_size = 1
|
|
global websocket, voices, offline_msg_done
|
|
|
|
for i in range(chunk_begin, chunk_begin + chunk_size):
|
|
offline_msg_done = False
|
|
voices = Queue()
|
|
|
|
if args.ssl == 1:
|
|
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_CLIENT)
|
|
ssl_context.check_hostname = False
|
|
ssl_context.verify_mode = ssl.CERT_NONE
|
|
uri = "wss://{}:{}".format(args.host, args.port)
|
|
else:
|
|
uri = "ws://{}:{}".format(args.host, args.port)
|
|
ssl_context = None
|
|
|
|
print("connect to", uri)
|
|
|
|
async with websockets.connect(
|
|
uri, subprotocols=["binary"], ping_interval=None, ssl=ssl_context
|
|
) as websocket:
|
|
meeting_tag = f"{id}_{i}"
|
|
writer = MeetingWriter(args.log_dir, meeting_id=meeting_tag, flush_every=args.log_flush_every)
|
|
try:
|
|
if args.audio_in is not None:
|
|
task = asyncio.create_task(record_from_scp(i, 1))
|
|
else:
|
|
task = asyncio.create_task(record_microphone())
|
|
task3 = asyncio.create_task(message(str(id) + "_" + str(i), writer)) # processid+fileid
|
|
await asyncio.gather(task, task3)
|
|
finally:
|
|
writer.close()
|
|
|
|
return
|
|
|
|
|
|
def one_thread(id, chunk_begin, chunk_size):
|
|
# ✅ 子进程里用 asyncio.run 更稳
|
|
asyncio.run(ws_client(id, chunk_begin, chunk_size))
|
|
|
|
|
|
if __name__ == "__main__":
|
|
# for microphone
|
|
if args.audio_in is None:
|
|
p = Process(target=one_thread, args=(0, 0, 0))
|
|
p.start()
|
|
p.join()
|
|
print("end")
|
|
else:
|
|
# calculate the number of wavs for each process
|
|
if args.audio_in.endswith(".scp"):
|
|
f_scp = open(args.audio_in)
|
|
wavs = f_scp.readlines()
|
|
else:
|
|
wavs = [args.audio_in]
|
|
|
|
total_len = len(wavs)
|
|
if total_len >= args.thread_num:
|
|
chunk_size = int(total_len / args.thread_num)
|
|
remain_wavs = total_len - chunk_size * args.thread_num
|
|
else:
|
|
chunk_size = 1
|
|
remain_wavs = 0
|
|
|
|
process_list = []
|
|
chunk_begin = 0
|
|
for i in range(args.thread_num):
|
|
now_chunk_size = chunk_size
|
|
if remain_wavs > 0:
|
|
now_chunk_size = chunk_size + 1
|
|
remain_wavs = remain_wavs - 1
|
|
|
|
p = Process(target=one_thread, args=(i, chunk_begin, now_chunk_size))
|
|
chunk_begin = chunk_begin + now_chunk_size
|
|
p.start()
|
|
process_list.append(p)
|
|
|
|
for p in process_list:
|
|
p.join()
|
|
|
|
print("end")
|