mirror of
https://github.com/modelscope/FunASR
synced 2025-09-15 14:48:36 +08:00
275 lines
8.7 KiB
Python
275 lines
8.7 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 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("--chunk_interval", type=int, default=60, help="chunk")
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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("--asr_prompt", type=str, default="Copy:", help="asr prompt")
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parser.add_argument("--s2tt_prompt", type=str, default="Translate the following sentence into English:", help="s2tt prompt")
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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("--ssl", type=int, default=1, help="1 for ssl connect, 0 for no ssl")
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parser.add_argument("--mode", type=str, default="online", 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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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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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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"wav_name": "microphone",
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"is_speaking": True,
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"asr_prompt": args.asr_prompt,
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"s2tt_prompt": args.s2tt_prompt,
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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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sample_rate = args.audio_fs
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wav_format = "pcm"
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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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"wav_name": "microphone",
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"is_speaking": True,
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"asr_prompt": args.asr_prompt,
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"s2tt_prompt": args.s2tt_prompt,
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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 # 60 * args.chunk_size[1] / args.chunk_interval / 1000
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sleep_duration = 60 * args.chunk_size[1] / args.chunk_interval / 1000
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await asyncio.sleep(sleep_duration)
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await asyncio.sleep(2)
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await websocket.close()
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async def message(id):
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global websocket, voices
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text_print = ""
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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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asr_text = meg["asr_text"]
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s2tt_text = meg["s2tt_text"]
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text_print = "\n\n" + "ASR: " + asr_text + "\n\n" + "S2TT: " + s2tt_text
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os.system("clear")
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print("\rpid" + str(id) + ": " + text_print)
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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
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for i in range(chunk_begin, chunk_begin + chunk_size):
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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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