FunASR/funasr/runtime/python/websocket/wss_client_asr.py

# -*- encoding: utf-8 -*-
import os
import time
import websockets, ssl
import asyncio
# import threading
import argparse
import json
import traceback
from multiprocessing import Process
from funasr.fileio.datadir_writer import DatadirWriter

import logging

SUPPORT_AUDIO_TYPE_SETS = ['.wav', '.pcm']
logging.basicConfig(level=logging.ERROR)

parser = argparse.ArgumentParser()
parser.add_argument("--host",
                    type=str,
                    default="localhost",
                    required=False,
                    help="host ip, localhost, 0.0.0.0")
parser.add_argument("--port",
                    type=int,
                    default=10095,
                    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=10,
                    help="chunk")
parser.add_argument("--audio_in",
                    type=str,
                    default=None,
                    help="audio_in")
parser.add_argument("--send_without_sleep",
                    action="store_true",
                    default=False,
                    help="if audio_in is set, send_without_sleep")
parser.add_argument("--test_thread_num",
                    type=int,
                    default=1,
                    help="test_thread_num")
parser.add_argument("--words_max_print",
                    type=int,
                    default=10000,
                    help="chunk")
parser.add_argument("--output_dir",
                    type=str,
                    default=None,
                    help="output_dir")

parser.add_argument("--ssl",
                    type=int,
                    default=1,
                    help="1 for ssl connect, 0 for no ssl")
parser.add_argument("--mode",
                    type=str,
                    default="2pass",
                    help="offline, online, 2pass")

args = parser.parse_args()
args.chunk_size = [int(x) for x in args.chunk_size.split(",")]
print(args)
# voices = asyncio.Queue()
from queue import Queue

voices = Queue()
ibest_writer = None
if args.output_dir is not None:
    writer = DatadirWriter(args.output_dir)
    ibest_writer = writer[f"1best_recog"]


async def record_microphone():
    is_finished = False
    import pyaudio
    # print("2")
    global voices
    FORMAT = pyaudio.paInt16
    CHANNELS = 1
    RATE = 16000
    chunk_size = 60 * args.chunk_size[1] / args.chunk_interval
    CHUNK = int(RATE / 1000 * chunk_size)

    p = pyaudio.PyAudio()

    stream = p.open(format=FORMAT,
                    channels=CHANNELS,
                    rate=RATE,
                    input=True,
                    frames_per_buffer=CHUNK)

    message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
                          "wav_name": "microphone", "is_speaking": True})
    voices.put(message)
    while True:
        data = stream.read(CHUNK)
        message = data
        voices.put(message)
        await asyncio.sleep(0.005)

async def record_from_scp(chunk_begin, chunk_size):
    global voices
    is_finished = False
    if args.audio_in.endswith(".scp"):
        f_scp = open(args.audio_in)
        wavs = f_scp.readlines()
    else:
        wavs = [args.audio_in]
    if chunk_size > 0:
        wavs = wavs[chunk_begin:chunk_begin + chunk_size]
    for wav in wavs:
        wav_splits = wav.strip().split()
 
        wav_name = wav_splits[0] if len(wav_splits) > 1 else "demo"
        wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
        if not len(wav_path.strip())>0:
           continue
        if wav_path.endswith(".pcm"):
            with open(wav_path, "rb") as f:
                audio_bytes = f.read()
        elif wav_path.endswith(".wav"):
            import wave
            with wave.open(wav_path, "rb") as wav_file:
                params = wav_file.getparams()
                frames = wav_file.readframes(wav_file.getnframes())
                audio_bytes = bytes(frames)
        else:
            raise NotImplementedError(
                f'Not supported audio type')

        # 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)

        # send first time
        message = json.dumps({"mode": args.mode, "chunk_size": args.chunk_size, "chunk_interval": args.chunk_interval,
                              "wav_name": wav_name, "is_speaking": True})
        #voices.put(message)
        await websocket.send(message)
        is_speaking = True
        for i in range(chunk_num):

            beg = i * stride
            data = audio_bytes[beg:beg + stride]
            message = data
            #voices.put(message)
            await websocket.send(message)
            if i == chunk_num - 1:
                is_speaking = False
                message = json.dumps({"is_speaking": is_speaking})
                #voices.put(message)
                await websocket.send(message)
            # print("data_chunk: ", len(data_chunk))
            # print(voices.qsize())
            sleep_duration = 0.001 if args.send_without_sleep else 60 * args.chunk_size[1] / args.chunk_interval / 1000
            await asyncio.sleep(sleep_duration)
    while not voices.empty():
         await asyncio.sleep(1)
    await asyncio.sleep(3)
    await websocket.close()
     
 
 

 
             
async def message(id):
    global websocket,voices
    text_print = ""
    text_print_2pass_online = ""
    text_print_2pass_offline = ""
    try:
       while True:
        
            meg = await websocket.recv()
            meg = json.loads(meg)
            wav_name = meg.get("wav_name", "demo")
            # print(wav_name)
            text = meg["text"]
            if ibest_writer is not None:
                ibest_writer["text"][wav_name] = text

            if meg["mode"] == "online":
                text_print += "{}".format(text)
                text_print = text_print[-args.words_max_print:]
                os.system('clear')
                print("\rpid" + str(id) + ": " + text_print)
            elif meg["mode"] == "offline":
                text_print += "{}".format(text)
                text_print = text_print[-args.words_max_print:]
                os.system('clear')
                print("\rpid" + str(id) + ": " + text_print)
            else:
                if meg["mode"] == "2pass-online":
                    text_print_2pass_online += "{}".format(text)
                    text_print = text_print_2pass_offline + text_print_2pass_online
                else:
                    text_print_2pass_online = ""
                    text_print = text_print_2pass_offline + "{}".format(text)
                    text_print_2pass_offline += "{}".format(text)
                text_print = text_print[-args.words_max_print:]
                os.system('clear')
                print("\rpid" + str(id) + ": " + text_print)

    except Exception as e:
            print("Exception:", e)
            #traceback.print_exc()
            #await websocket.close()
 


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)
            #traceback.print_exc()
            exit(0)

async def ws_client(id, chunk_begin, chunk_size):
  if args.audio_in is None:
       chunk_begin=0
       chunk_size=1
  global websocket,voices
  for i in range(chunk_begin,chunk_begin+chunk_size):
    voices = Queue()
    if args.ssl == 1:
        ssl_context = ssl.SSLContext()
        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:
        if args.audio_in is not None:
            task = asyncio.create_task(record_from_scp(i, 1))
        else:
            task = asyncio.create_task(record_microphone())
        #task2 = asyncio.create_task(ws_send())
        task3 = asyncio.create_task(message(id))
        await asyncio.gather(task, task3)
  exit(0)
    

def one_thread(id, chunk_begin, chunk_size):
    asyncio.get_event_loop().run_until_complete(ws_client(id, chunk_begin, chunk_size))
    asyncio.get_event_loop().run_forever()

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 preocess
        if args.audio_in.endswith(".scp"):
            f_scp = open(args.audio_in)
            wavs = f_scp.readlines()
        else:
            wavs = [args.audio_in]
        for wav in wavs:
            wav_splits = wav.strip().split()
            wav_name = wav_splits[0] if len(wav_splits) > 1 else "demo"
            wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
            audio_type = os.path.splitext(wav_path)[-1].lower()
            if audio_type not in SUPPORT_AUDIO_TYPE_SETS:
                raise NotImplementedError(
                    f'Not supported audio type: {audio_type}')

        total_len = len(wavs)
        if total_len >= args.test_thread_num:
            chunk_size = int(total_len / args.test_thread_num)
            remain_wavs = total_len - chunk_size * args.test_thread_num
        else:
            chunk_size = 1
            remain_wavs = 0

        process_list = []
        chunk_begin = 0
        for i in range(args.test_thread_num):
            now_chunk_size = chunk_size
            if remain_wavs > 0:
                now_chunk_size = chunk_size + 1
                remain_wavs = remain_wavs - 1
            # process i handle wavs at chunk_begin and size of now_chunk_size
            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 i in process_list:
            p.join()

        print('end')