websocket offline

游雁

2023-05-07 296ec00997d4ad4715286c8dad9cc2226d064b71

websocket offline

 funasr/runtime/python/websocket/README.md

@@ -22,15 +22,13 @@

### Start server
#### ASR offline server
```shell
python ws_server_offline.py --port 10095 --asr_model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch"
```

[//]: # (```shell)

[//]: # (python ws_server_online.py --host "0.0.0.0" --port 10095 --asr_model "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch")

[//]: # (```)
#### ASR streaming server
```shell
python ws_server_online.py --host "0.0.0.0" --port 10095 --asr_model_online "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
python ws_server_online.py --port 10095 --asr_model_online "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online"
```

#### ASR offline/online 2pass server
@@ -51,17 +49,31 @@
```

### Start client
#### Recording from mircrophone
#### ASR offline client
##### Recording from mircrophone
```shell
# --chunk_interval, "10": 600/10=60ms, "5"=600/5=120ms, "20": 600/12=30ms
python ws_client.py --host "0.0.0.0" --port 10095 --chunk_interval 10 --words_max_print 100
```
##### Loadding from wav.scp(kaldi style)
```shell
# --chunk_interval, "10": 600/10=60ms, "5"=600/5=120ms, "20": 600/12=30ms
python ws_client.py --host "0.0.0.0" --port 10095 --chunk_interval 10 --words_max_print 100 --audio_in "./data/wav.scp" --send_without_sleep
```
#### ASR streaming client
##### Recording from mircrophone
```shell
# --chunk_size, "5,10,5"=600ms, "8,8,4"=480ms
python ws_client.py --host "127.0.0.1" --port 10095 --chunk_size "5,10,5" --words_max_print 100
python ws_client.py --host "0.0.0.0" --port 10095 --chunk_size "5,10,5" --words_max_print 100
```
#### Loadding from wav.scp(kaldi style)
##### Loadding from wav.scp(kaldi style)
```shell
# --chunk_size, "5,10,5"=600ms, "8,8,4"=480ms
python ws_client.py --host "127.0.0.1" --port 10095 --chunk_size "5,10,5" --audio_in "./data/wav.scp" --words_max_print 100
python ws_client.py --host "0.0.0.0" --port 10095 --chunk_size "5,10,5" --audio_in "./data/wav.scp" --words_max_print 100
```

#### ASR offline/online 2pass client

## Acknowledge
1. This project is maintained by [FunASR community](https://github.com/alibaba-damo-academy/FunASR).
2. We acknowledge [zhaoming](https://github.com/zhaomingwork/FunASR/tree/fix_bug_for_python_websocket) for contributing the websocket service.

 funasr/runtime/python/websocket/parse_args.py

@@ -31,5 +31,10 @@
                    type=int,
                    default=1,
                    help="0 for cpu, 1 for gpu")
parser.add_argument("--ncpu",
                    type=int,
                    default=1,
                    help="cpu cores")

args = parser.parse_args()
print(args)

 funasr/runtime/python/websocket/ws_client.py

@@ -31,7 +31,10 @@
                    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,
@@ -43,12 +46,11 @@

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()

# 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
@@ -75,11 +77,9 @@
        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())

        await asyncio.sleep(0.005)

# 其他函数可以通过调用send(data)来发送数据，例如：
async def record_from_scp():
    import wave
    global voices
@@ -95,15 +95,11 @@
        # bytes_f = open(wav_path, "rb")
        # bytes_data = bytes_f.read()
        with wave.open(wav_path, "rb") as wav_file:
            # 获取音频参数
            params = wav_file.getparams()
            # 获取头信息的长度
            # header_length = wav_file.getheaders()[0][1]
            # 读取音频帧数据，跳过头信息
            # wav_file.setpos(header_length)
            frames = wav_file.readframes(wav_file.getnframes())

        # 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(60*args.chunk_size[1]/args.chunk_interval/1000*16000*2)
@@ -120,8 +116,8 @@
            voices.put(message)
            # print("data_chunk: ", len(data_chunk))
            # print(voices.qsize())
        
            await asyncio.sleep(60*args.chunk_size[1]/args.chunk_interval/1000)
            sleep_duration = 0.001 if args.send_without_sleep else 60*args.chunk_size[1]/args.chunk_interval/1000
            await asyncio.sleep(sleep_duration)

    is_finished = True
    message = json.dumps({"is_finished": is_finished})
@@ -136,7 +132,7 @@
            data = voices.get()
            voices.task_done()
            try:
                await websocket.send(data) # 通过ws对象发送数据
                await websocket.send(data)
            except Exception as e:
                print('Exception occurred:', e)
                traceback.print_exc()
@@ -155,9 +151,14 @@
            meg = json.loads(meg)
            # print(meg, end = '')
            # print("\r")
            text_print += " {}".format(meg["text"][0]) 
            # print(meg)
            text = meg["text"][0]
            if meg["mode"] == "online":
                text_print += " {}".format(text)
            else:
                text_print += "{}".format(text)
            text_print = text_print[-args.words_max_print:]
            #os.system('clear')
            os.system('clear')
            print("\r"+str(id)+":"+text_print)
        except Exception as e:
            print("Exception:", e)
@@ -177,17 +178,15 @@
            exit(0)

async def ws_client(id):
    global websocket # 定义一个全局变量ws，用于保存websocket连接对象
    # uri = "ws://11.167.134.197:8899"
    global websocket
    uri = "ws://{}:{}".format(args.host, args.port)
    #ws = await websockets.connect(uri, subprotocols=["binary"]) # 创建一个长连接
    async for websocket in websockets.connect(uri, subprotocols=["binary"], ping_interval=None):
        if args.audio_in is not None:
            task = asyncio.create_task(record_from_scp()) # 创建一个后台任务录音
            task = asyncio.create_task(record_from_scp())
        else:
            task = asyncio.create_task(record_microphone())  # 创建一个后台任务录音
        task2 = asyncio.create_task(ws_send()) # 创建一个后台任务发送
        task3 = asyncio.create_task(message(id)) # 创建一个后台接收消息的任务
            task = asyncio.create_task(record_microphone())
        task2 = asyncio.create_task(ws_send())
        task3 = asyncio.create_task(message(id))
        await asyncio.gather(task, task2, task3)

def one_thread(id):
@@ -198,13 +197,13 @@
if __name__ == '__main__':
    process_list = []
    for i in range(args.test_thread_num):   
        p = Process(target=one_thread,args=(i,)) #实例化进程对象
        p = Process(target=one_thread,args=(i,))
        p.start()
        process_list.append(p)

    for i in process_list:
        p.join()

    print('结束测试')
    print('end')
 


 funasr/runtime/python/websocket/ws_server_offline.py

New file
@@ -0,0 +1,147 @@
import asyncio
import json
import websockets
import time
import logging
import tracemalloc
import numpy as np

from parse_args import args
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
from funasr.runtime.python.onnxruntime.funasr_onnx.utils.frontend import load_bytes

tracemalloc.start()

logger = get_logger(log_level=logging.CRITICAL)
logger.setLevel(logging.CRITICAL)


websocket_users = set()

print("model loading")
# asr
inference_pipeline_asr = pipeline(
    task=Tasks.auto_speech_recognition,
    model=args.asr_model,
    ngpu=args.ngpu,
    ncpu=args.ncpu,
    model_revision=None)


# vad
inference_pipeline_vad = pipeline(
    task=Tasks.voice_activity_detection,
    model=args.vad_model,
    model_revision=None,
    output_dir=None,
    batch_size=1,
    mode='online',
    ngpu=args.ngpu,
    ncpu=args.ncpu,
)

if args.punc_model != "":
    inference_pipeline_punc = pipeline(
        task=Tasks.punctuation,
        model=args.punc_model,
        model_revision=None,
        ngpu=args.ngpu,
        ncpu=args.ncpu,
    )
else:
    inference_pipeline_punc = None

print("model loaded")

async def ws_serve(websocket, path):
    frames = []
    frames_asr = []
    global websocket_users
    websocket_users.add(websocket)
    websocket.param_dict_asr = {}
    websocket.param_dict_vad = {'in_cache': dict(), "is_final": False}
    websocket.param_dict_punc = {'cache': list()}
    websocket.vad_pre_idx = 0
    speech_start = False

    try:
        async for message in websocket:
            message = json.loads(message)
            is_finished = message["is_finished"]
            if not is_finished:
                audio = bytes(message['audio'], 'ISO-8859-1')
                frames.append(audio)
                duration_ms = len(audio)//32
                websocket.vad_pre_idx += duration_ms

                is_speaking = message["is_speaking"]
                websocket.param_dict_vad["is_final"] = not is_speaking
                if speech_start:
                    frames_asr.append(audio)
                speech_start_i, speech_end_i = await async_vad(websocket, audio)
                if speech_start_i:
                    speech_start = True
                    beg_bias = (websocket.vad_pre_idx-speech_start_i)//duration_ms
                    frames_pre = frames[-beg_bias:]
                    frames_asr = []
                    frames_asr.extend(frames_pre)
                if speech_end_i or not is_speaking:
                    audio_in = b"".join(frames_asr)
                    await async_asr(websocket, audio_in)
                    frames_asr = []
                    speech_start = False
                    if not is_speaking:
                        websocket.vad_pre_idx = 0
                        frames = []
                    else:
                        frames = frames[-10:]

     
    except websockets.ConnectionClosed:
        print("ConnectionClosed...", websocket_users)
        websocket_users.remove(websocket)
    except websockets.InvalidState:
        print("InvalidState...")
    except Exception as e:
        print("Exception:", e)


async def async_vad(websocket, audio_in):

    segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)

    speech_start = False
    speech_end = False
    
    if len(segments_result) == 0 or len(segments_result["text"]) > 1:
        return speech_start, speech_end
    if segments_result["text"][0][0] != -1:
        speech_start = segments_result["text"][0][0]
    if segments_result["text"][0][1] != -1:
        speech_end = True
    return speech_start, speech_end


async def async_asr(websocket, audio_in):
            if len(audio_in) > 0:
                # print(len(audio_in))
                audio_in = load_bytes(audio_in)
                
                rec_result = inference_pipeline_asr(audio_in=audio_in,
                                                    param_dict=websocket.param_dict_asr)
                # print(rec_result)
                if inference_pipeline_punc is not None and 'text' in rec_result and len(rec_result["text"])>0:
                    rec_result = inference_pipeline_punc(text_in=rec_result['text'],
                                                         param_dict=websocket.param_dict_punc)
                    # print(rec_result)
                    message = json.dumps({"mode": "offline", "text": [rec_result["text"]]})
                    await websocket.send(message)
                    
 


start_server = websockets.serve(ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None)
asyncio.get_event_loop().run_until_complete(start_server)
asyncio.get_event_loop().run_forever()

 funasr/runtime/python/websocket/ws_server_online.py

@@ -12,7 +12,7 @@
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
from funasr_onnx.utils.frontend import load_bytes
from funasr.runtime.python.onnxruntime.funasr_onnx.utils.frontend import load_bytes

tracemalloc.start()

@@ -28,6 +28,8 @@
inference_pipeline_asr_online = pipeline(
    task=Tasks.auto_speech_recognition,
    model=args.asr_model_online,
    ngpu=args.ngpu,
    ncpu=args.ncpu,
    model_revision='v1.0.4')

print("model loaded")
@@ -63,14 +65,14 @@

     
    except websockets.ConnectionClosed:
        print("ConnectionClosed...", websocket_users)    # 链接断开
        print("ConnectionClosed...", websocket_users)
        websocket_users.remove(websocket)
    except websockets.InvalidState:
        print("InvalidState...")    # 无效状态
        print("InvalidState...")
    except Exception as e:
        print("Exception:", e)
 
async def async_asr_online(websocket,audio_in): # ASR推理
async def async_asr_online(websocket,audio_in):
            if len(audio_in) > 0:
                audio_in = load_bytes(audio_in)
                rec_result = inference_pipeline_asr_online(audio_in=audio_in,
@@ -83,7 +85,6 @@
                            rec_result["text"][0]=rec_result["text"][0] #.replace(" ","")
                        message = json.dumps({"mode": "online", "text": rec_result["text"]})
                        await websocket.send(message)