游雁
2023-04-27 ba8d73d57db031fa7a1265d2c837ff694d5c5c93
funasr/runtime/python/websocket/ws_client.py
File was renamed from funasr/runtime/python/websocket/ASR_client.py
@@ -1,4 +1,5 @@
# -*- encoding: utf-8 -*-
import os
import time
import websockets
import asyncio
@@ -18,29 +19,36 @@
                    required=False,
                    help="grpc server port")
parser.add_argument("--chunk_size",
                    type=str,
                    default="5, 10, 5",
                    help="chunk")
parser.add_argument("--chunk_interval",
                    type=int,
                    default=300,
                    help="ms")
                    default=10,
                    help="chunk")
parser.add_argument("--audio_in",
                    type=str,
                    default=None,
                    help="audio_in")
args = parser.parse_args()
args.chunk_size = [int(x) for x in args.chunk_size.split(",")]
# voices = asyncio.Queue()
from queue import Queue
voices = Queue()
# 其他函数可以通过调用send(data)来发送数据,例如:
async def record_microphone():
    is_finished = False
    import pyaudio
    #print("2")
    global voices 
    FORMAT = pyaudio.paInt16
    CHANNELS = 1
    RATE = 16000
    CHUNK = int(RATE / 1000 * args.chunk_size)
    chunk_size = 60*args.chunk_size[1]/args.chunk_interval
    CHUNK = int(RATE / 1000 * chunk_size)
    p = pyaudio.PyAudio()
@@ -54,7 +62,7 @@
        data = stream.read(CHUNK)
        data = data.decode('ISO-8859-1')
        message = json.dumps({"chunk": args.chunk_size, "is_speaking": is_speaking, "audio": data})
        message = json.dumps({"chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval, "audio": data, "is_speaking": is_speaking, "is_finished": is_finished})
        
        voices.put(message)
        #print(voices.qsize())
@@ -65,6 +73,7 @@
async def record_from_scp():
    import wave
    global voices
    is_finished = False
    if args.audio_in.endswith(".scp"):
        f_scp = open(args.audio_in)
        wavs = f_scp.readlines()
@@ -86,9 +95,10 @@
        # 将音频帧数据转换为字节类型的数据
        audio_bytes = bytes(frames)
        stride = int(args.chunk_size/1000*16000*2)
        # stride = int(args.chunk_size/1000*16000*2)
        stride = int(60*args.chunk_size[1]/args.chunk_interval/1000*16000*2)
        chunk_num = (len(audio_bytes)-1)//stride + 1
        print(stride)
        # print(stride)
        is_speaking = True
        for i in range(chunk_num):
            if i == chunk_num-1:
@@ -96,13 +106,16 @@
            beg = i*stride
            data = audio_bytes[beg:beg+stride]
            data = data.decode('ISO-8859-1')
            message = json.dumps({"chunk": args.chunk_size, "is_speaking": is_speaking, "audio": data})
            message = json.dumps({"chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval, "is_speaking": is_speaking, "audio": data, "is_finished": is_finished})
            voices.put(message)
            # print("data_chunk: ", len(data_chunk))
            # print(voices.qsize())
        
            await asyncio.sleep(args.chunk_size/1000)
            await asyncio.sleep(60*args.chunk_size[1]/args.chunk_interval/1000)
    is_finished = True
    message = json.dumps({"is_finished": is_finished})
    voices.put(message)
async def ws_send():
    global voices
@@ -123,14 +136,31 @@
async def message():
    global websocket
    text_print = ""
    while True:
        try:
            meg = await websocket.recv()
            meg = json.loads(meg)
            # print(meg, end = '')
            # print("\r")
            text = meg["text"][0]
            text_print += text
            text_print = text_print[-55:]
            os.system('clear')
            print("\r"+text_print)
        except Exception as e:
            print("Exception:", e)
async def print_messge():
    global websocket
    while True:
        try:
            meg = await websocket.recv()
            meg = json.loads(meg)
            print(meg)
        except Exception as e:
            print("Exception:", e)
            print("Exception:", e)
async def ws_client():