Merge pull request #1771 from lym0302/add_streaming_cli
[server] add streaming tts demospull/1776/head
TianYuan
2 years ago
committed by
GitHub
commit
f256bb9c0e
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# This is the parameter configuration file for PaddleSpeech Serving.
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#################################################################################
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# SERVER SETTING #
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#################################################################################
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host: 127.0.0.1
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port: 8092
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# The task format in the engin_list is: <speech task>_<engine type>
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# engine_list choices = ['tts_online', 'tts_online-onnx']
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# protocol = ['websocket', 'http'] (only one can be selected).
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protocol: 'http'
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engine_list: ['tts_online-onnx']
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#################################################################################
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# ENGINE CONFIG #
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#################################################################################
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################################### TTS #########################################
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################### speech task: tts; engine_type: online #######################
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tts_online:
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# am (acoustic model) choices=['fastspeech2_csmsc', 'fastspeech2_cnndecoder_csmsc']
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am: 'fastspeech2_csmsc'
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am_config:
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am_ckpt:
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am_stat:
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phones_dict:
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tones_dict:
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speaker_dict:
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spk_id: 0
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# voc (vocoder) choices=['mb_melgan_csmsc, hifigan_csmsc']
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voc: 'mb_melgan_csmsc'
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voc_config:
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voc_ckpt:
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voc_stat:
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# others
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lang: 'zh'
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device: 'cpu' # set 'gpu:id' or 'cpu'
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am_block: 42
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am_pad: 12
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voc_block: 14
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voc_pad: 14
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#################################################################################
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# ENGINE CONFIG #
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#################################################################################
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################################### TTS #########################################
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################### speech task: tts; engine_type: online-onnx #######################
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tts_online-onnx:
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# am (acoustic model) choices=['fastspeech2_csmsc_onnx', 'fastspeech2_cnndecoder_csmsc_onnx']
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am: 'fastspeech2_cnndecoder_csmsc_onnx'
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# am_ckpt is a list, if am is fastspeech2_cnndecoder_csmsc_onnx, am_ckpt = [encoder model, decoder model, postnet model];
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# if am is fastspeech2_csmsc_onnx, am_ckpt = [ckpt model];
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am_ckpt: # list
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am_stat:
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phones_dict:
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tones_dict:
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speaker_dict:
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spk_id: 0
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am_sample_rate: 24000
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am_sess_conf:
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device: "cpu" # set 'gpu:id' or 'cpu'
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use_trt: False
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cpu_threads: 4
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# voc (vocoder) choices=['mb_melgan_csmsc_onnx, hifigan_csmsc_onnx']
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voc: 'hifigan_csmsc_onnx'
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voc_ckpt:
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voc_sample_rate: 24000
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voc_sess_conf:
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device: "cpu" # set 'gpu:id' or 'cpu'
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use_trt: False
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cpu_threads: 4
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# others
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lang: 'zh'
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am_block: 42
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am_pad: 12
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voc_block: 14
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voc_pad: 14
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voc_upsample: 300
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@ -0,0 +1,3 @@
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#!/bin/bash
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# start server
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paddlespeech_server start --config_file ./conf/tts_online_application.yaml
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@ -0,0 +1,7 @@
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#!/bin/bash
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# http client test
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paddlespeech_client tts --server_ip 127.0.0.1 --port 8092 --protocol http --input "您好，欢迎使用百度飞桨语音合成服务。" --output output.wav
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# websocket client test
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#paddlespeech_client tts --server_ip 127.0.0.1 --port 8092 --protocol websocket --input "您好，欢迎使用百度飞桨语音合成服务。" --output output.wav
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@ -1,112 +0,0 @@
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# 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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@ -1,160 +0,0 @@
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# Copyright (c) 2022 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 _thread as thread
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import argparse
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import base64
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import json
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import ssl
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import threading
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import time
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import pyaudio
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import websocket
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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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flag = 1
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st = 0.0
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all_bytes = 0.0
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class WsParam(object):
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# 初始化
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def __init__(self, text, server="127.0.0.1", port=8090):
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self.server = server
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self.port = port
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self.url = "ws://" + self.server + ":" + str(self.port) + "/ws/tts"
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self.text = text
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# 生成url
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def create_url(self):
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return self.url
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def play_audio():
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global stream
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global buffer
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while True:
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time.sleep(0.05)
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if not buffer: # buffer 为空
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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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t = threading.Thread(target=play_audio)
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def on_message(ws, message):
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global flag
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global t
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global buffer
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global st
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global all_bytes
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try:
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message = json.loads(message)
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audio = message["audio"]
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audio = base64.b64decode(audio) # bytes
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status = message["status"]
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all_bytes += len(audio)
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if status == 0:
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print("create successfully.")
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elif status == 1:
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mutex.acquire()
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buffer += audio
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mutex.release()
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if flag:
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print(f"首包响应：{time.time() - st} s")
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flag = 0
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print("Start playing audio")
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t.start()
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elif status == 2:
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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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print("ws is closed")
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ws.close()
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else:
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print("infer error")
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except Exception as e:
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print("receive msg,but parse exception:", e)
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# 收到websocket错误的处理
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def on_error(ws, error):
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print("### error:", error)
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# 收到websocket关闭的处理
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def on_close(ws):
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print("### closed ###")
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# 收到websocket连接建立的处理
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def on_open(ws):
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def run(*args):
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global st
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text_base64 = str(
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base64.b64encode((wsParam.text).encode('utf-8')), "UTF8")
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d = {"text": text_base64}
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d = json.dumps(d)
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print("Start sending text data")
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st = time.time()
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ws.send(d)
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thread.start_new_thread(run, ())
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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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help="A sentence to be synthesized",
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default="您好，欢迎使用语音合成服务。")
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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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print("***************************************")
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print("Server ip: ", args.server)
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print("Server port: ", args.port)
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print("Sentence to be synthesized: ", args.text)
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print("***************************************")
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wsParam = WsParam(text=args.text, server=args.server, port=args.port)
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websocket.enableTrace(False)
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wsUrl = wsParam.create_url()
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ws = websocket.WebSocketApp(
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wsUrl, on_message=on_message, on_error=on_error, on_close=on_close)
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ws.on_open = on_open
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ws.run_forever(sslopt={"cert_reqs": ssl.CERT_NONE})
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t.join()
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print("End of playing audio")
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stream.stop_stream()
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stream.close()
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p.terminate()
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