From 28ccfbfc51068a663a80764e14074df5edf2b5ba Mon Sep 17 00:00:00 2001
From: kongdeqiang <kongdeqiang960204@163.com>
Date: 星期五, 13 三月 2026 17:41:41 +0800
Subject: [PATCH] 提交

---
 runtime/python/websocket/funasr_wss_server.py |  555 +++++++++++++++++++++++++++++--------------------------
 1 files changed, 294 insertions(+), 261 deletions(-)

diff --git a/runtime/python/websocket/funasr_wss_server.py b/runtime/python/websocket/funasr_wss_server.py
index fa82ea5..1f957a9 100644
--- a/runtime/python/websocket/funasr_wss_server.py
+++ b/runtime/python/websocket/funasr_wss_server.py
@@ -10,71 +10,56 @@
 
 
 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(
+    "--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")
+parser.add_argument(
+    "--keyfile",
+    type=str,
+    default="../../ssl_key/server.key",
+    required=False,
+    help="keyfile for ssl",
+)
 args = parser.parse_args()
 
 
@@ -84,229 +69,277 @@
 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,
-                      )
+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,
-                                )
+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,
-                      )
+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,
-	                       )
+    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
-
+    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 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()
-
+    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:
-					websocket.status_dict_asr_online["chunk_size"] = messagejson["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)
+    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 "hotwords" in messagejson:
+                    websocket.status_dict_asr["hotword"] = messagejson["hotwords"]
+                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
+
+    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)
+    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)
 
+    else:
+        mode = "2pass-offline" if "2pass" in websocket.mode else websocket.mode
+        message = json.dumps(
+            {
+                "mode": mode,
+                "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(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)
+
+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)
+    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()

--
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