From 219c2482ab755fbd4e49dfbdee91bf1a8a4ec49a Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期五, 19 五月 2023 11:33:27 +0800
Subject: [PATCH] websocket 2pass bugfix
---
funasr/runtime/python/websocket/ws_server_2pass.py | 91 ++++++++++++++++++++++++++-------------------
1 files changed, 53 insertions(+), 38 deletions(-)
diff --git a/funasr/runtime/python/websocket/ws_server_2pass.py b/funasr/runtime/python/websocket/ws_server_2pass.py
index ced67ff..e5cab9c 100644
--- a/funasr/runtime/python/websocket/ws_server_2pass.py
+++ b/funasr/runtime/python/websocket/ws_server_2pass.py
@@ -46,7 +46,7 @@
inference_pipeline_punc = pipeline(
task=Tasks.punctuation,
model=args.punc_model,
- model_revision=None,
+ model_revision="v1.0.2",
ngpu=args.ngpu,
ncpu=args.ncpu,
)
@@ -74,47 +74,58 @@
websocket.param_dict_punc = {'cache': list()}
websocket.vad_pre_idx = 0
speech_start = False
+ speech_end_i = False
+ websocket.wav_name = "microphone"
+ print("new user connected", flush=True)
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
- websocket.param_dict_asr_online["is_final"] = not is_speaking
- websocket.param_dict_asr_online["chunk_size"] = message["chunk_size"]
- websocket.wav_name = message.get("wav_name", "demo")
- # asr online
- frames_asr_online.append(audio)
- if len(frames_asr_online) % message["chunk_interval"] == 0:
- audio_in = b"".join(frames_asr_online)
- await async_asr_online(websocket, audio_in)
- frames_asr_online = []
- if speech_start:
- frames_asr.append(audio)
- # vad online
- 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 isinstance(message, str):
+ messagejson = json.loads(message)
+
+ if "is_speaking" in messagejson:
+ websocket.is_speaking = messagejson["is_speaking"]
+ websocket.param_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:
+ websocket.param_dict_asr_online["chunk_size"] = messagejson["chunk_size"]
+ 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.param_dict_asr_online["is_final"] = speech_end_i
+ if len(frames_asr_online) % websocket.chunk_interval == 0 or websocket.param_dict_asr_online["is_final"]:
+
+ audio_in = b"".join(frames_asr_online)
+ await async_asr_online(websocket, audio_in)
+ frames_asr_online = []
+ if speech_start:
+ frames_asr.append(message)
+ # vad online
+ speech_start_i, speech_end_i = await async_vad(websocket, message)
+ 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)
# asr punc offline
- if speech_end_i or not is_speaking:
+ if speech_end_i or not websocket.is_speaking:
+ # print("vad end point")
audio_in = b"".join(frames_asr)
await async_asr(websocket, audio_in)
frames_asr = []
speech_start = False
- frames_asr_online = []
- websocket.param_dict_asr_online = {"cache": dict()}
- if not is_speaking:
+ # frames_asr_online = []
+ # websocket.param_dict_asr_online = {"cache": dict()}
+ if not websocket.is_speaking:
websocket.vad_pre_idx = 0
frames = []
websocket.param_dict_vad = {'in_cache': dict()}
@@ -159,17 +170,21 @@
rec_result = inference_pipeline_punc(text_in=rec_result['text'],
param_dict=websocket.param_dict_punc)
# print("offline", rec_result)
- message = json.dumps({"mode": "2pass-offline", "text": rec_result["text"], "wav_name": websocket.wav_name})
- await websocket.send(message)
+ if 'text' in rec_result:
+ message = json.dumps({"mode": "2pass-offline", "text": rec_result["text"], "wav_name": websocket.wav_name})
+ await websocket.send(message)
async def async_asr_online(websocket, audio_in):
if len(audio_in) > 0:
audio_in = load_bytes(audio_in)
+ # print(websocket.param_dict_asr_online.get("is_final", False))
rec_result = inference_pipeline_asr_online(audio_in=audio_in,
param_dict=websocket.param_dict_asr_online)
- if websocket.param_dict_asr_online["is_final"]:
- websocket.param_dict_asr_online["cache"] = dict()
+ # print(rec_result)
+ if websocket.param_dict_asr_online.get("is_final", False):
+ return
+ # websocket.param_dict_asr_online["cache"] = dict()
if "text" in rec_result:
if rec_result["text"] != "sil" and rec_result["text"] != "waiting_for_more_voice":
# print("online", rec_result)
--
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