# -*- 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=True, help="if audio_in is set, send_without_sleep") parser.add_argument("--thread_num", type=int, default=1, help="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() offline_msg_done=False 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) await websocket.send(message) while True: data = stream.read(CHUNK) message = data #voices.put(message) await websocket.send(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) sleep_duration = 0.001 if args.mode == "offline" else 60 * args.chunk_size[1] / args.chunk_interval / 1000 await asyncio.sleep(sleep_duration) # when all data sent, we need to close websocket while not voices.empty(): await asyncio.sleep(1) await asyncio.sleep(3) # offline model need to wait for message recved if args.mode=="offline": global offline_msg_done while not offline_msg_done: await asyncio.sleep(1) await websocket.close() async def message(id): global websocket,voices,offline_msg_done 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") 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) offline_msg_done=True 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) offline_msg_done=True 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,offline_msg_done for i in range(chunk_begin,chunk_begin+chunk_size): offline_msg_done=False 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()) task3 = asyncio.create_task(message(str(id)+"_"+str(i))) #processid+fileid 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.thread_num: chunk_size = int(total_len / args.thread_num) remain_wavs = total_len - chunk_size * args.thread_num else: chunk_size = 1 remain_wavs = 0 process_list = [] chunk_begin = 0 for i in range(args.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')