From 4ee715e70e36cdba7b05fe044fecab9cf4fa16ff Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期一, 03 七月 2023 17:23:02 +0800
Subject: [PATCH] websocket bug
---
funasr/runtime/python/websocket/wss_srv_asr.py | 22 ++++++++++------------
1 files changed, 10 insertions(+), 12 deletions(-)
diff --git a/funasr/runtime/python/websocket/wss_srv_asr.py b/funasr/runtime/python/websocket/wss_srv_asr.py
index 948619b..09f2305 100644
--- a/funasr/runtime/python/websocket/wss_srv_asr.py
+++ b/funasr/runtime/python/websocket/wss_srv_asr.py
@@ -35,8 +35,6 @@
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,
@@ -69,9 +67,9 @@
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)
+ # 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()
@@ -95,7 +93,7 @@
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)
@@ -124,7 +122,7 @@
# 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)
@@ -134,14 +132,14 @@
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)
@@ -172,15 +170,15 @@
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
--
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