From 3360a1d9453ef0ce441cc41b0090d09b3bb296bb Mon Sep 17 00:00:00 2001
From: aky15 <ankeyuthu@gmail.com>
Date: 星期二, 04 七月 2023 20:02:45 +0800
Subject: [PATCH] Dev bat (#701)
---
funasr/runtime/python/websocket/wss_srv_asr.py | 96 +++++++++++++++++++++++++++++++++++++++++-------
1 files changed, 82 insertions(+), 14 deletions(-)
diff --git a/funasr/runtime/python/websocket/wss_srv_asr.py b/funasr/runtime/python/websocket/wss_srv_asr.py
index 71c97e6..fd039ae 100644
--- a/funasr/runtime/python/websocket/wss_srv_asr.py
+++ b/funasr/runtime/python/websocket/wss_srv_asr.py
@@ -5,8 +5,8 @@
import logging
import tracemalloc
import numpy as np
+import argparse
import ssl
-from parse_args import args
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
from modelscope.utils.logger import get_logger
@@ -16,6 +16,54 @@
logger = get_logger(log_level=logging.CRITICAL)
logger.setLevel(logging.CRITICAL)
+
+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="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch",
+ help="model from modelscope")
+parser.add_argument("--asr_model_online",
+ type=str,
+ default="damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online",
+ help="model from modelscope")
+parser.add_argument("--vad_model",
+ type=str,
+ default="damo/speech_fsmn_vad_zh-cn-16k-common-pytorch",
+ help="model from modelscope")
+parser.add_argument("--punc_model",
+ type=str,
+ default="damo/punc_ct-transformer_zh-cn-common-vad_realtime-vocab272727",
+ help="model from modelscope")
+parser.add_argument("--ngpu",
+ type=int,
+ default=1,
+ help="0 for cpu, 1 for gpu")
+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()
@@ -35,8 +83,6 @@
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,
@@ -58,15 +104,36 @@
model=args.asr_model_online,
ngpu=args.ngpu,
ncpu=args.ncpu,
- model_revision='v1.0.4')
+ model_revision='v1.0.4',
+ update_model='v1.0.4',
+ mode='paraformer_streaming')
-print("model loaded")
+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.param_dict_asr_online = {"cache": dict()}
+ websocket.param_dict_vad = {'in_cache': dict(), "is_final": True}
+ websocket.param_dict_asr_online["is_final"]=True
+ # audio_in=b''.join(np.zeros(int(16000),dtype=np.int16))
+ # inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)
+ # inference_pipeline_asr_online(audio_in=audio_in, param_dict=websocket.param_dict_asr_online)
+ 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.param_dict_asr = {}
websocket.param_dict_asr_online = {"cache": dict()}
@@ -74,7 +141,7 @@
websocket.param_dict_punc = {'cache': list()}
websocket.vad_pre_idx = 0
speech_start = False
- speech_end_i = False
+ speech_end_i = -1
websocket.wav_name = "microphone"
websocket.mode = "2pass"
print("new user connected", flush=True)
@@ -103,7 +170,7 @@
# asr online
frames_asr_online.append(message)
- websocket.param_dict_asr_online["is_final"] = speech_end_i
+ websocket.param_dict_asr_online["is_final"] = speech_end_i != -1
if len(frames_asr_online) % websocket.chunk_interval == 0 or websocket.param_dict_asr_online["is_final"]:
if websocket.mode == "2pass" or websocket.mode == "online":
audio_in = b"".join(frames_asr_online)
@@ -113,14 +180,14 @@
frames_asr.append(message)
# vad online
speech_start_i, speech_end_i = await async_vad(websocket, message)
- if speech_start_i:
+ 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 or not websocket.is_speaking:
+ 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)
@@ -138,7 +205,8 @@
except websockets.ConnectionClosed:
- print("ConnectionClosed...", websocket_users)
+ print("ConnectionClosed...", websocket_users,flush=True)
+ await ws_reset(websocket)
websocket_users.remove(websocket)
except websockets.InvalidState:
print("InvalidState...")
@@ -150,15 +218,15 @@
segments_result = inference_pipeline_vad(audio_in=audio_in, param_dict=websocket.param_dict_vad)
- speech_start = False
- speech_end = False
+ speech_start = -1
+ speech_end = -1
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
+ speech_end = segments_result["text"][0][1]
return speech_start, speech_end
@@ -207,4 +275,4 @@
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()
\ No newline at end of file
+asyncio.get_event_loop().run_forever()
--
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