FunASR/runtime/python/websocket/funasr_wss_server.py

335 lines
12 KiB
Python

import asyncio
import json
import websockets
import time
import logging
import tracemalloc
import numpy as np
import argparse
import ssl
parser = argparse.ArgumentParser()
parser.add_argument(
"--host", type=str, default="0.0.0.0", 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(
"--asr_model",
type=str,
default="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
help="model from modelscope",
)
parser.add_argument("--asr_model_revision", type=str, default="v2.0.4", help="")
parser.add_argument(
"--asr_model_online",
type=str,
default="iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online",
help="model from modelscope",
)
parser.add_argument("--asr_model_online_revision", type=str, default="v2.0.4", help="")
parser.add_argument(
"--vad_model",
type=str,
default="iic/speech_fsmn_vad_zh-cn-16k-common-pytorch",
help="model from modelscope",
)
parser.add_argument("--vad_model_revision", type=str, default="v2.0.4", help="")
parser.add_argument(
"--punc_model",
type=str,
default="iic/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
help="model from modelscope",
)
parser.add_argument("--punc_model_revision", type=str, default="v2.0.4", help="")
parser.add_argument("--ngpu", type=int, default=1, help="0 for cpu, 1 for gpu")
parser.add_argument("--device", type=str, default="cuda", help="cuda, cpu")
parser.add_argument("--ncpu", type=int, default=4, help="cpu cores")
parser.add_argument(
"--certfile",
type=str,
default="../../ssl_key/server.crt",
required=False,
help="certfile for ssl",
)
parser.add_argument(
"--keyfile",
type=str,
default="../../ssl_key/server.key",
required=False,
help="keyfile for ssl",
)
args = parser.parse_args()
websocket_users = set()
print("model loading")
from funasr import AutoModel
# asr
model_asr = AutoModel(
model=args.asr_model,
model_revision=args.asr_model_revision,
ngpu=args.ngpu,
ncpu=args.ncpu,
device=args.device,
disable_pbar=True,
disable_log=True,
)
# asr
model_asr_streaming = AutoModel(
model=args.asr_model_online,
model_revision=args.asr_model_online_revision,
ngpu=args.ngpu,
ncpu=args.ncpu,
device=args.device,
disable_pbar=True,
disable_log=True,
)
# vad
model_vad = AutoModel(
model=args.vad_model,
model_revision=args.vad_model_revision,
ngpu=args.ngpu,
ncpu=args.ncpu,
device=args.device,
disable_pbar=True,
disable_log=True,
# chunk_size=60,
)
if args.punc_model != "":
model_punc = AutoModel(
model=args.punc_model,
model_revision=args.punc_model_revision,
ngpu=args.ngpu,
ncpu=args.ncpu,
device=args.device,
disable_pbar=True,
disable_log=True,
)
else:
model_punc = None
print("model loaded! only support one client at the same time now!!!!")
async def ws_reset(websocket):
print("ws reset now, total num is ", len(websocket_users))
websocket.status_dict_asr_online["cache"] = {}
websocket.status_dict_asr_online["is_final"] = True
websocket.status_dict_vad["cache"] = {}
websocket.status_dict_vad["is_final"] = True
websocket.status_dict_punc["cache"] = {}
await websocket.close()
async def clear_websocket():
for websocket in websocket_users:
await ws_reset(websocket)
websocket_users.clear()
async def ws_serve(websocket, path):
frames = []
frames_asr = []
frames_asr_online = []
global websocket_users
# await clear_websocket()
websocket_users.add(websocket)
websocket.status_dict_asr = {}
websocket.status_dict_asr_online = {"cache": {}, "is_final": False}
websocket.status_dict_vad = {"cache": {}, "is_final": False}
websocket.status_dict_punc = {"cache": {}}
websocket.chunk_interval = 10
websocket.vad_pre_idx = 0
speech_start = False
speech_end_i = -1
websocket.wav_name = "microphone"
websocket.mode = "2pass"
print("new user connected", flush=True)
try:
async for message in websocket:
if isinstance(message, str):
messagejson = json.loads(message)
if "is_speaking" in messagejson:
websocket.is_speaking = messagejson["is_speaking"]
websocket.status_dict_asr_online["is_final"] = not websocket.is_speaking
if "chunk_interval" in messagejson:
websocket.chunk_interval = messagejson["chunk_interval"]
if "wav_name" in messagejson:
websocket.wav_name = messagejson.get("wav_name")
if "chunk_size" in messagejson:
chunk_size = messagejson["chunk_size"]
if isinstance(chunk_size, str):
chunk_size = chunk_size.split(",")
websocket.status_dict_asr_online["chunk_size"] = [int(x) for x in chunk_size]
if "encoder_chunk_look_back" in messagejson:
websocket.status_dict_asr_online["encoder_chunk_look_back"] = messagejson[
"encoder_chunk_look_back"
]
if "decoder_chunk_look_back" in messagejson:
websocket.status_dict_asr_online["decoder_chunk_look_back"] = messagejson[
"decoder_chunk_look_back"
]
if "hotword" in messagejson:
websocket.status_dict_asr["hotword"] = messagejson["hotword"]
if "mode" in messagejson:
websocket.mode = messagejson["mode"]
websocket.status_dict_vad["chunk_size"] = int(
websocket.status_dict_asr_online["chunk_size"][1] * 60 / websocket.chunk_interval
)
if len(frames_asr_online) > 0 or len(frames_asr) > 0 or not isinstance(message, str):
if not isinstance(message, str):
frames.append(message)
duration_ms = len(message) // 32
websocket.vad_pre_idx += duration_ms
# asr online
frames_asr_online.append(message)
websocket.status_dict_asr_online["is_final"] = speech_end_i != -1
if (
len(frames_asr_online) % websocket.chunk_interval == 0
or websocket.status_dict_asr_online["is_final"]
):
if websocket.mode == "2pass" or websocket.mode == "online":
audio_in = b"".join(frames_asr_online)
try:
await async_asr_online(websocket, audio_in)
except:
print(f"error in asr streaming, {websocket.status_dict_asr_online}")
frames_asr_online = []
if speech_start:
frames_asr.append(message)
# vad online
try:
speech_start_i, speech_end_i = await async_vad(websocket, message)
except:
print("error in vad")
if speech_start_i != -1:
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)
# asr punc offline
if speech_end_i != -1 or not websocket.is_speaking:
# print("vad end point")
if websocket.mode == "2pass" or websocket.mode == "offline":
audio_in = b"".join(frames_asr)
try:
await async_asr(websocket, audio_in)
except:
print("error in asr offline")
frames_asr = []
speech_start = False
frames_asr_online = []
websocket.status_dict_asr_online["cache"] = {}
if not websocket.is_speaking:
websocket.vad_pre_idx = 0
frames = []
websocket.status_dict_vad["cache"] = {}
else:
frames = frames[-20:]
except websockets.ConnectionClosed:
print("ConnectionClosed...", websocket_users, flush=True)
await ws_reset(websocket)
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 = model_vad.generate(input=audio_in, **websocket.status_dict_vad)[0]["value"]
# print(segments_result)
speech_start = -1
speech_end = -1
if len(segments_result) == 0 or len(segments_result) > 1:
return speech_start, speech_end
if segments_result[0][0] != -1:
speech_start = segments_result[0][0]
if segments_result[0][1] != -1:
speech_end = segments_result[0][1]
return speech_start, speech_end
async def async_asr(websocket, audio_in):
if len(audio_in) > 0:
# print(len(audio_in))
rec_result = model_asr.generate(input=audio_in, **websocket.status_dict_asr)[0]
# print("offline_asr, ", rec_result)
if model_punc is not None and len(rec_result["text"]) > 0:
# print("offline, before punc", rec_result, "cache", websocket.status_dict_punc)
rec_result = model_punc.generate(
input=rec_result["text"], **websocket.status_dict_punc
)[0]
# print("offline, after punc", rec_result)
if len(rec_result["text"]) > 0:
# print("offline", rec_result)
mode = "2pass-offline" if "2pass" in websocket.mode else websocket.mode
message = json.dumps(
{
"mode": mode,
"text": rec_result["text"],
"wav_name": websocket.wav_name,
"is_final": websocket.is_speaking,
}
)
await websocket.send(message)
async def async_asr_online(websocket, audio_in):
if len(audio_in) > 0:
# print(websocket.status_dict_asr_online.get("is_final", False))
rec_result = model_asr_streaming.generate(
input=audio_in, **websocket.status_dict_asr_online
)[0]
# print("online, ", rec_result)
if websocket.mode == "2pass" and websocket.status_dict_asr_online.get("is_final", False):
return
# websocket.status_dict_asr_online["cache"] = dict()
if len(rec_result["text"]):
mode = "2pass-online" if "2pass" in websocket.mode else websocket.mode
message = json.dumps(
{
"mode": mode,
"text": rec_result["text"],
"wav_name": websocket.wav_name,
"is_final": websocket.is_speaking,
}
)
await websocket.send(message)
if len(args.certfile) > 0:
ssl_context = ssl.SSLContext(ssl.PROTOCOL_TLS_SERVER)
# Generate with Lets Encrypt, copied to this location, chown to current user and 400 permissions
ssl_cert = args.certfile
ssl_key = args.keyfile
ssl_context.load_cert_chain(ssl_cert, keyfile=ssl_key)
start_server = websockets.serve(
ws_serve, args.host, args.port, subprotocols=["binary"], ping_interval=None, ssl=ssl_context
)
else:
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()