# -*- 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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SUPPORT_AUDIO_TYPE_SETS = ['.wav', '.pcm']
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logging.basicConfig(level=logging.ERROR)
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parser = argparse.ArgumentParser()
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parser.add_argument("--host",
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type=str,
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default="localhost",
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required=False,
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help="host ip, localhost, 0.0.0.0")
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parser.add_argument("--port",
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type=int,
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default=10095,
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required=False,
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help="grpc server port")
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parser.add_argument("--chunk_size",
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type=str,
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default="5, 10, 5",
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help="chunk")
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parser.add_argument("--chunk_interval",
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type=int,
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default=10,
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help="chunk")
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parser.add_argument("--audio_in",
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type=str,
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default=None,
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help="audio_in")
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parser.add_argument("--send_without_sleep",
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action="store_true",
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default=False,
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help="if audio_in is set, send_without_sleep")
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parser.add_argument("--test_thread_num",
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type=int,
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default=1,
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help="test_thread_num")
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parser.add_argument("--words_max_print",
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type=int,
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default=10000,
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help="chunk")
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parser.add_argument("--output_dir",
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type=str,
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default=None,
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help="output_dir")
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parser.add_argument("--ssl",
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type=int,
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default=1,
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help="1 for ssl connect, 0 for no ssl")
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parser.add_argument("--mode",
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type=str,
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default="2pass",
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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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ibest_writer = None
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if args.output_dir is not None:
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writer = DatadirWriter(args.output_dir)
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ibest_writer = writer[f"1best_recog"]
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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(format=FORMAT,
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channels=CHANNELS,
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rate=RATE,
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input=True,
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frames_per_buffer=CHUNK)
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message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
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"wav_name": "microphone", "is_speaking": True})
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voices.put(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 asyncio.sleep(0.005)
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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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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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frames = wav_file.readframes(wav_file.getnframes())
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audio_bytes = bytes(frames)
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else:
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raise NotImplementedError(
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f'Not supported audio type')
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# stride = int(args.chunk_size/1000*16000*2)
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stride = int(60 * args.chunk_size[1] / args.chunk_interval / 1000 * 16000 * 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({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
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"wav_name": wav_name, "is_speaking": True})
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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 = 0.001 if args.send_without_sleep else 60 * args.chunk_size[1] / args.chunk_interval / 1000
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await asyncio.sleep(sleep_duration)
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# when all data sent, we need to close websocket
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while not voices.empty():
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await asyncio.sleep(1)
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await asyncio.sleep(3)
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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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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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if ibest_writer is not None:
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ibest_writer["text"][wav_name] = text
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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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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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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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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 print_messge():
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global websocket
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while True:
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try:
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meg = await websocket.recv()
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meg = json.loads(meg)
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print(meg)
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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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exit(0)
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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(uri, subprotocols=["binary"], ping_interval=None, ssl=ssl_context) 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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#task2 = asyncio.create_task(ws_send())
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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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if audio_type not in SUPPORT_AUDIO_TYPE_SETS:
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raise NotImplementedError(
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f'Not supported audio type: {audio_type}')
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total_len = len(wavs)
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if total_len >= args.test_thread_num:
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chunk_size = int(total_len / args.test_thread_num)
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remain_wavs = total_len - chunk_size * args.test_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.test_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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