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https://github.com/modelscope/FunASR
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wss llm
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游雁
parent
f2af56b678
commit
260d037d55
394
runtime/python/websocket/funasr_wss_client_llm.py
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394
runtime/python/websocket/funasr_wss_client_llm.py
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@ -0,0 +1,394 @@
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# -*- 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 threading
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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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# from funasr.fileio.datadir_writer import DatadirWriter
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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("--audio_in", type=str, default=None, help="audio_in")
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parser.add_argument("--audio_fs", type=int, default=16000, help="audio_fs")
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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=True,
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help="if audio_in is set, send_without_sleep",
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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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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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# voices = asyncio.Queue()
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from queue import Queue
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voices = Queue()
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offline_msg_done = False
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if args.output_dir is not None:
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# if os.path.exists(args.output_dir):
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# os.remove(args.output_dir)
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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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is_finished = False
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import pyaudio
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# print("2")
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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)
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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)
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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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)
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# voices.put(message)
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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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message = data
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# voices.put(message)
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await websocket.send(message)
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await asyncio.sleep(0.0005)
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async def record_from_scp(chunk_begin, chunk_size):
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global voices
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is_finished = False
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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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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)
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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)
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else:
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hotword_msg = args.hotword
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print(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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params = wav_file.getparams()
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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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# print(stride)
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# send first time
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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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)
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# voices.put(message)
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await websocket.send(message)
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is_speaking = True
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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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message = data
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# voices.put(message)
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await websocket.send(message)
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if i == chunk_num - 1:
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is_speaking = False
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message = json.dumps({"is_speaking": is_speaking})
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# voices.put(message)
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await websocket.send(message)
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sleep_duration = 0.00001
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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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# offline model need to wait for message recved
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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 websocket.close()
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async def message(id):
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global websocket, voices, offline_msg_done
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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["text"]
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timestamp = ""
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offline_msg_done = meg.get("is_final", False)
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if "timestamp" in meg:
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timestamp = meg["timestamp"]
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if ibest_writer is not None:
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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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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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os.system("clear")
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print("\rpid" + 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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# text_print = text_print[-args.words_max_print:]
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# os.system('clear')
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print("\rpid" + str(id) + ": " + wav_name + ": " + text_print)
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offline_msg_done = True
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else:
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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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text_print = text_print[-args.words_max_print :]
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os.system("clear")
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print("\rpid" + str(id) + ": " + text_print)
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# offline_msg_done=True
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except Exception as e:
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print("Exception:", e)
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# traceback.print_exc()
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# await websocket.close()
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async def ws_client(id, chunk_begin, chunk_size):
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if args.audio_in is None:
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chunk_begin = 0
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chunk_size = 1
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global websocket, voices, offline_msg_done
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for i in range(chunk_begin, chunk_begin + chunk_size):
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offline_msg_done = False
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voices = Queue()
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if args.ssl == 1:
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ssl_context = ssl.SSLContext()
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ssl_context.check_hostname = False
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ssl_context.verify_mode = ssl.CERT_NONE
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uri = "wss://{}:{}".format(args.host, args.port)
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else:
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uri = "ws://{}:{}".format(args.host, args.port)
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ssl_context = None
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print("connect to", uri)
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async with websockets.connect(
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uri, subprotocols=["binary"], ping_interval=None, ssl=ssl_context
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) as websocket:
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if args.audio_in is not None:
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task = asyncio.create_task(record_from_scp(i, 1))
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else:
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task = asyncio.create_task(record_microphone())
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task3 = asyncio.create_task(message(str(id) + "_" + str(i))) # processid+fileid
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await asyncio.gather(task, task3)
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exit(0)
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def one_thread(id, chunk_begin, chunk_size):
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asyncio.get_event_loop().run_until_complete(ws_client(id, chunk_begin, chunk_size))
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asyncio.get_event_loop().run_forever()
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if __name__ == "__main__":
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# for microphone
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if args.audio_in is None:
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p = Process(target=one_thread, args=(0, 0, 0))
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p.start()
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p.join()
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print("end")
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else:
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# calculate the number of wavs for each preocess
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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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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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audio_type = os.path.splitext(wav_path)[-1].lower()
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total_len = len(wavs)
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if total_len >= args.thread_num:
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chunk_size = int(total_len / args.thread_num)
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remain_wavs = total_len - chunk_size * args.thread_num
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else:
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chunk_size = 1
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remain_wavs = 0
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process_list = []
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chunk_begin = 0
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for i in range(args.thread_num):
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now_chunk_size = chunk_size
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if remain_wavs > 0:
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now_chunk_size = chunk_size + 1
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remain_wavs = remain_wavs - 1
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# process i handle wavs at chunk_begin and size of now_chunk_size
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p = Process(target=one_thread, args=(i, chunk_begin, now_chunk_size))
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chunk_begin = chunk_begin + now_chunk_size
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p.start()
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process_list.append(p)
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for i in process_list:
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p.join()
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print("end")
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"""
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python funasr_wss_client.py --host "127.0.0.1" --port 10095 --audio_in audio_file
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"""
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@ -192,6 +192,7 @@ async def model_inference(
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history=None,
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text_usr="",
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):
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beg0 = time.time()
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if his_state is None:
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his_state = model_dict
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model = his_state["model"]
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@ -243,7 +244,9 @@ async def model_inference(
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beg_llm = time.time()
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for new_text in streamer:
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end_llm = time.time()
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print(f"generated new text： {new_text}, time: {end_llm - beg_llm:.2f}")
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print(
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f"generated new text： {new_text}, time_fr_receive: {end_llm - beg0:.2f}, time_llm_decode: {end_llm - beg_llm:.2f}"
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)
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if len(new_text) > 0:
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res += new_text.replace("<|im_end|>", "")
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