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PaddleSpeech/paddlespeech/server/tests/tts/online/ws_client_playaudio.py

161 lines
4.3 KiB

# Copyright (c) 2022 PaddlePaddle Authors. All Rights Reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import _thread as thread
import argparse
import base64
import json
import ssl
import threading
import time
import pyaudio
import websocket
mutex = threading.Lock()
buffer = b''
p = pyaudio.PyAudio()
stream = p.open(
format=p.get_format_from_width(2), channels=1, rate=24000, output=True)
flag = 1
st = 0.0
all_bytes = 0.0
class Ws_Param(object):
# 初始化
def __init__(self, text, server="127.0.0.1", port=8090):
self.server = server
self.port = port
self.url = "ws://" + self.server + ":" + str(self.port) + "/ws/tts"
self.text = text
# 生成url
def create_url(self):
return self.url
def play_audio():
global stream
global buffer
while True:
time.sleep(0.05)
if not buffer: # buffer 为空
break
mutex.acquire()
stream.write(buffer)
buffer = b''
mutex.release()
t = threading.Thread(target=play_audio)
def on_message(ws, message):
global flag
global t
global buffer
global st
global all_bytes
try:
message = json.loads(message)
audio = message["audio"]
audio = base64.b64decode(audio) # bytes
status = message["status"]
all_bytes += len(audio)
if status == 0:
print("create successfully.")
elif status == 1:
mutex.acquire()
buffer += audio
mutex.release()
if flag:
print(f"首包响应:{time.time() - st} s")
flag = 0
print("Start playing audio")
t.start()
elif status == 2:
final_response = time.time() - st
duration = all_bytes / 2 / 24000
print(f"尾包响应:{final_response} s")
print(f"音频时长:{duration} s")
print(f"RTF: {final_response / duration}")
print("ws is closed")
ws.close()
else:
print("infer error")
except Exception as e:
print("receive msg,but parse exception:", e)
# 收到websocket错误的处理
def on_error(ws, error):
print("### error:", error)
# 收到websocket关闭的处理
def on_close(ws):
print("### closed ###")
# 收到websocket连接建立的处理
def on_open(ws):
def run(*args):
global st
text_base64 = str(
base64.b64encode((wsParam.text).encode('utf-8')), "UTF8")
d = {"text": text_base64}
d = json.dumps(d)
print("Start sending text data")
st = time.time()
ws.send(d)
thread.start_new_thread(run, ())
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument(
"--text",
type=str,
help="A sentence to be synthesized",
default="您好,欢迎使用语音合成服务。")
parser.add_argument(
"--server", type=str, help="server ip", default="127.0.0.1")
parser.add_argument("--port", type=int, help="server port", default=8092)
args = parser.parse_args()
print("***************************************")
print("Server ip: ", args.server)
print("Server port: ", args.port)
print("Sentence to be synthesized: ", args.text)
print("***************************************")
wsParam = Ws_Param(text=args.text, server=args.server, port=args.port)
websocket.enableTrace(False)
wsUrl = wsParam.create_url()
ws = websocket.WebSocketApp(
wsUrl, on_message=on_message, on_error=on_error, on_close=on_close)
ws.on_open = on_open
ws.run_forever(sslopt={"cert_reqs": ssl.CERT_NONE})
t.join()
print("End of playing audio")
stream.stop_stream()
stream.close()
p.terminate()