From 678a6c0f7293a86fb1046cf043afec29e88fd5f1 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期一, 24 四月 2023 15:54:54 +0800
Subject: [PATCH] websocket
---
funasr/runtime/python/websocket/ASR_client.py | 41 ++++++++++++++++++++++++++++++++++-------
funasr/runtime/python/websocket/ASR_server_2pass.py | 2 +-
2 files changed, 35 insertions(+), 8 deletions(-)
diff --git a/funasr/runtime/python/websocket/ASR_client.py b/funasr/runtime/python/websocket/ASR_client.py
index fa95328..cc0e7b6 100644
--- a/funasr/runtime/python/websocket/ASR_client.py
+++ b/funasr/runtime/python/websocket/ASR_client.py
@@ -1,9 +1,8 @@
-import pyaudio
+
# import websocket #鍖哄埆鏈嶅姟绔繖閲屾槸 websocket-client搴�
import time
import websockets
import asyncio
-from queue import Queue
# import threading
import argparse
import json
@@ -30,12 +29,13 @@
args = parser.parse_args()
+# voices = asyncio.Queue()
+from queue import Queue
voices = Queue()
-
-
# 鍏朵粬鍑芥暟鍙互閫氳繃璋冪敤send(data)鏉ュ彂閫佹暟鎹紝渚嬪锛�
-async def record():
+async def record_microphone():
+ import pyaudio
#print("2")
global voices
FORMAT = pyaudio.paInt16
@@ -59,8 +59,32 @@
#print(voices.qsize())
await asyncio.sleep(0.01)
-
+# 鍏朵粬鍑芥暟鍙互閫氳繃璋冪敤send(data)鏉ュ彂閫佹暟鎹紝渚嬪锛�
+async def record_from_scp():
+ global voices
+ if args.audio_in.endswith(".scp"):
+ f_scp = open(args.audio_in)
+ wavs = f_scp.readlines()
+ else:
+ wavs = [args.audio_in]
+ for wav in wavs:
+ wav_splits = wav.strip().split()
+ wav_path = wav_splits[1] if len(wav_splits) > 1 else wav_splits[0]
+ bytes = open(wav_path, "rb")
+ bytes = bytes.read()
+
+ stride = int(args.chunk_size/1000*16000*2)
+ chunk_num = (len(bytes)-1)//stride + 1
+ for i in range(chunk_num):
+ beg = i*stride
+ data_chunk = bytes[beg:beg+stride]
+ voices.put(data_chunk)
+ # print("data_chunk: ", len(data_chunk))
+ # print(voices.qsize())
+
+ await asyncio.sleep(args.chunk_size/1000)
+
async def ws_send():
global voices
@@ -97,7 +121,10 @@
uri = "ws://{}:{}".format(args.host, args.port)
#ws = await websockets.connect(uri, subprotocols=["binary"]) # 鍒涘缓涓�涓暱杩炴帴
async for websocket in websockets.connect(uri, subprotocols=["binary"], ping_interval=None):
- task = asyncio.create_task(record()) # 鍒涘缓涓�涓悗鍙颁换鍔″綍闊�
+ if args.audio_in is not None:
+ task = asyncio.create_task(record_from_scp()) # 鍒涘缓涓�涓悗鍙颁换鍔″綍闊�
+ else:
+ task = asyncio.create_task(record_microphone()) # 鍒涘缓涓�涓悗鍙颁换鍔″綍闊�
task2 = asyncio.create_task(ws_send()) # 鍒涘缓涓�涓悗鍙颁换鍔″彂閫�
task3 = asyncio.create_task(message()) # 鍒涘缓涓�涓悗鍙版帴鏀舵秷鎭殑浠诲姟
await asyncio.gather(task, task2, task3)
diff --git a/funasr/runtime/python/websocket/ASR_server_2pass.py b/funasr/runtime/python/websocket/ASR_server_2pass.py
index 55dc2e2..135a3cc 100644
--- a/funasr/runtime/python/websocket/ASR_server_2pass.py
+++ b/funasr/runtime/python/websocket/ASR_server_2pass.py
@@ -105,7 +105,7 @@
inference_pipeline_asr_online = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer_asr_nat-zh-cn-16k-common-vocab8404-online',
- model_revision='v1.0.2')
+ model_revision=None)
print("model loaded")
--
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