Merge branch 'main' of https://github.com/alibaba-damo-academy/FunASR

nichongjia-2007

2023-07-07 1ce704d8c09bd4d4c7e5ab087f951f31fad9fca6

 funasr/runtime/python/websocket/funasr_wss_client.py

File was renamed from funasr/runtime/python/websocket/wss_client_asr.py
@@ -8,11 +8,10 @@
import json
import traceback
from multiprocessing import Process
from funasr.fileio.datadir_writer import DatadirWriter
# from funasr.fileio.datadir_writer import DatadirWriter

import logging

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

parser = argparse.ArgumentParser()
@@ -42,10 +41,10 @@
                    action="store_true",
                    default=True,
                    help="if audio_in is set, send_without_sleep")
parser.add_argument("--test_thread_num",
parser.add_argument("--thread_num",
                    type=int,
                    default=1,
                    help="test_thread_num")
                    help="thread_num")
parser.add_argument("--words_max_print",
                    type=int,
                    default=10000,
@@ -72,11 +71,13 @@

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"]
    # if os.path.exists(args.output_dir):
    #     os.remove(args.output_dir)
        
    if not os.path.exists(args.output_dir):
        os.makedirs(args.output_dir)


async def record_microphone():
@@ -100,11 +101,13 @@

    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)
    #voices.put(message)
    await websocket.send(message)
    while True:
        data = stream.read(CHUNK)
        message = data
        voices.put(message)
        #voices.put(message)
        await websocket.send(message)
        await asyncio.sleep(0.005)

async def record_from_scp(chunk_begin, chunk_size):
@@ -134,8 +137,17 @@
                frames = wav_file.readframes(wav_file.getnframes())
                audio_bytes = bytes(frames)
        else:
            raise NotImplementedError(
                f'Not supported audio type')
            import ffmpeg
            try:
                # This launches a subprocess to decode audio while down-mixing and resampling as necessary.
                # Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
                audio_bytes, _ = (
                    ffmpeg.input(wav_path, threads=0)
                    .output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=16000)
                    .run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
                )
            except ffmpeg.Error as e:
                raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e

        # stride = int(args.chunk_size/1000*16000*2)
        stride = int(60 * args.chunk_size[1] / args.chunk_interval / 1000 * 16000 * 2)
@@ -164,10 +176,9 @@
            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)
    
    if not args.mode=="offline":
        await asyncio.sleep(2)
    # offline model need to wait for message recved
    
    if args.mode=="offline":
@@ -176,17 +187,18 @@
         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 = ""
    if args.output_dir is not None:
        ibest_writer = open(os.path.join(args.output_dir, "text.{}".format(id)), "a", encoding="utf-8")
    else:
        ibest_writer = None
    try:
       while True:
        
@@ -194,9 +206,11 @@
            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 ibest_writer is not None:
                text_write_line = "{}\t{}\n".format(wav_name, text)
                ibest_writer.write(text_write_line)
                
            if meg["mode"] == "online":
                text_print += "{}".format(text)
                text_print = text_print[-args.words_max_print:]
@@ -204,10 +218,10 @@
                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
                # text_print = text_print[-args.words_max_print:]
                # os.system('clear')
                print("\rpid" + str(id) + ": " + wav_name + ": " + text_print)
                offline_msg_done = True
            else:
                if meg["mode"] == "2pass-online":
                    text_print_2pass_online += "{}".format(text)
@@ -219,6 +233,7 @@
                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)
@@ -227,17 +242,6 @@
 


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:
@@ -262,7 +266,6 @@
            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(str(id)+"_"+str(i))) #processid+fileid
        await asyncio.gather(task, task3)
  exit(0)
@@ -291,21 +294,19 @@
            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
        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.test_thread_num):
        for i in range(args.thread_num):
            now_chunk_size = chunk_size
            if remain_wavs > 0:
                now_chunk_size = chunk_size + 1