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113 lines
3.2 KiB
113 lines
3.2 KiB
# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved.
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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import argparse
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import base64
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import json
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import threading
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import time
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import pyaudio
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import requests
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mutex = threading.Lock()
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buffer = b''
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p = pyaudio.PyAudio()
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stream = p.open(
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format=p.get_format_from_width(2), channels=1, rate=24000, output=True)
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max_fail = 50
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def play_audio():
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global stream
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global buffer
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global max_fail
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while True:
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if not buffer:
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max_fail -= 1
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time.sleep(0.05)
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if max_fail < 0:
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break
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mutex.acquire()
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stream.write(buffer)
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buffer = b''
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mutex.release()
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def test(args):
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global mutex
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global buffer
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params = {
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"text": args.text,
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"spk_id": args.spk_id,
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"speed": args.speed,
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"volume": args.volume,
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"sample_rate": args.sample_rate,
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"save_path": ''
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}
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all_bytes = 0.0
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t = threading.Thread(target=play_audio)
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flag = 1
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url = "http://" + str(args.server) + ":" + str(
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args.port) + "/paddlespeech/streaming/tts"
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st = time.time()
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html = requests.post(url, json.dumps(params), stream=True)
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for chunk in html.iter_content(chunk_size=1024):
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mutex.acquire()
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chunk = base64.b64decode(chunk) # bytes
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buffer += chunk
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mutex.release()
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if flag:
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first_response = time.time() - st
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print(f"首包响应：{first_response} s")
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flag = 0
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t.start()
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all_bytes += len(chunk)
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final_response = time.time() - st
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duration = all_bytes / 2 / 24000
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print(f"尾包响应：{final_response} s")
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print(f"音频时长：{duration} s")
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print(f"RTF: {final_response / duration}")
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t.join()
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stream.stop_stream()
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stream.close()
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p.terminate()
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument(
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'--text',
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type=str,
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default="您好，欢迎使用语音合成服务。",
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help='A sentence to be synthesized')
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parser.add_argument('--spk_id', type=int, default=0, help='Speaker id')
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parser.add_argument('--speed', type=float, default=1.0, help='Audio speed')
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parser.add_argument(
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'--volume', type=float, default=1.0, help='Audio volume')
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parser.add_argument(
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'--sample_rate',
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type=int,
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default=0,
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help='Sampling rate, the default is the same as the model')
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parser.add_argument(
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"--server", type=str, help="server ip", default="127.0.0.1")
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parser.add_argument("--port", type=int, help="server port", default=8092)
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args = parser.parse_args()
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test(args)
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