Hui Zhang
4 years ago
committed by
GitHub
commit
d65b63b28d
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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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# task choices = ['asr_online', 'tts_online']
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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']
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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']
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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']
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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: # 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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@ -0,0 +1,13 @@
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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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@ -0,0 +1,220 @@
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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 base64
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import time
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import numpy as np
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import paddle
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from paddlespeech.cli.log import logger
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from paddlespeech.cli.tts.infer import TTSExecutor
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from paddlespeech.server.engine.base_engine import BaseEngine
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from paddlespeech.server.utils.audio_process import float2pcm
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from paddlespeech.server.utils.util import get_chunks
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__all__ = ['TTSEngine']
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class TTSServerExecutor(TTSExecutor):
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def __init__(self):
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super().__init__()
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pass
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@paddle.no_grad()
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def infer(
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self,
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text: str,
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lang: str='zh',
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am: str='fastspeech2_csmsc',
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spk_id: int=0,
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am_block: int=42,
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am_pad: int=12,
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voc_block: int=14,
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voc_pad: int=14, ):
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"""
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Model inference and result stored in self.output.
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"""
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am_name = am[:am.rindex('_')]
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am_dataset = am[am.rindex('_') + 1:]
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get_tone_ids = False
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merge_sentences = False
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frontend_st = time.time()
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if lang == 'zh':
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input_ids = self.frontend.get_input_ids(
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text,
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merge_sentences=merge_sentences,
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get_tone_ids=get_tone_ids)
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phone_ids = input_ids["phone_ids"]
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if get_tone_ids:
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tone_ids = input_ids["tone_ids"]
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elif lang == 'en':
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input_ids = self.frontend.get_input_ids(
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text, merge_sentences=merge_sentences)
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phone_ids = input_ids["phone_ids"]
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else:
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print("lang should in {'zh', 'en'}!")
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self.frontend_time = time.time() - frontend_st
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for i in range(len(phone_ids)):
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am_st = time.time()
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part_phone_ids = phone_ids[i]
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# am
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if am_name == 'speedyspeech':
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part_tone_ids = tone_ids[i]
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mel = self.am_inference(part_phone_ids, part_tone_ids)
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# fastspeech2
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else:
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# multi speaker
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if am_dataset in {"aishell3", "vctk"}:
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mel = self.am_inference(
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part_phone_ids, spk_id=paddle.to_tensor(spk_id))
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else:
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mel = self.am_inference(part_phone_ids)
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am_et = time.time()
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# voc streaming
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voc_upsample = self.voc_config.n_shift
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mel_chunks = get_chunks(mel, voc_block, voc_pad, "voc")
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chunk_num = len(mel_chunks)
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voc_st = time.time()
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for i, mel_chunk in enumerate(mel_chunks):
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sub_wav = self.voc_inference(mel_chunk)
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front_pad = min(i * voc_block, voc_pad)
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if i == 0:
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sub_wav = sub_wav[:voc_block * voc_upsample]
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elif i == chunk_num - 1:
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sub_wav = sub_wav[front_pad * voc_upsample:]
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else:
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sub_wav = sub_wav[front_pad * voc_upsample:(
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front_pad + voc_block) * voc_upsample]
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yield sub_wav
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class TTSEngine(BaseEngine):
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"""TTS server engine
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Args:
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metaclass: Defaults to Singleton.
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"""
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def __init__(self, name=None):
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"""Initialize TTS server engine
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"""
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super(TTSEngine, self).__init__()
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def init(self, config: dict) -> bool:
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self.executor = TTSServerExecutor()
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self.config = config
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assert "fastspeech2_csmsc" in config.am and (
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config.voc == "hifigan_csmsc-zh" or config.voc == "mb_melgan_csmsc"
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), 'Please check config, am support: fastspeech2, voc support: hifigan_csmsc-zh or mb_melgan_csmsc.'
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try:
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if self.config.device:
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self.device = self.config.device
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else:
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self.device = paddle.get_device()
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paddle.set_device(self.device)
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except Exception as e:
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logger.error(
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"Set device failed, please check if device is already used and the parameter 'device' in the yaml file"
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)
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logger.error("Initialize TTS server engine Failed on device: %s." %
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(self.device))
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return False
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try:
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self.executor._init_from_path(
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am=self.config.am,
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am_config=self.config.am_config,
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am_ckpt=self.config.am_ckpt,
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am_stat=self.config.am_stat,
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phones_dict=self.config.phones_dict,
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tones_dict=self.config.tones_dict,
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speaker_dict=self.config.speaker_dict,
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voc=self.config.voc,
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voc_config=self.config.voc_config,
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voc_ckpt=self.config.voc_ckpt,
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voc_stat=self.config.voc_stat,
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lang=self.config.lang)
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except Exception as e:
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logger.error("Failed to get model related files.")
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logger.error("Initialize TTS server engine Failed on device: %s." %
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(self.device))
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return False
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self.am_block = self.config.am_block
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self.am_pad = self.config.am_pad
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self.voc_block = self.config.voc_block
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self.voc_pad = self.config.voc_pad
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logger.info("Initialize TTS server engine successfully on device: %s." %
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(self.device))
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return True
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def preprocess(self, text_bese64: str=None, text_bytes: bytes=None):
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# Convert byte to text
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if text_bese64:
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text_bytes = base64.b64decode(text_bese64) # base64 to bytes
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text = text_bytes.decode('utf-8') # bytes to text
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return text
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def run(self,
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sentence: str,
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spk_id: int=0,
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speed: float=1.0,
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volume: float=1.0,
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sample_rate: int=0,
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save_path: str=None):
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""" run include inference and postprocess.
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Args:
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sentence (str): text to be synthesized
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spk_id (int, optional): speaker id for multi-speaker speech synthesis. Defaults to 0.
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speed (float, optional): speed. Defaults to 1.0.
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volume (float, optional): volume. Defaults to 1.0.
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sample_rate (int, optional): target sample rate for synthesized audio,
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0 means the same as the model sampling rate. Defaults to 0.
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save_path (str, optional): The save path of the synthesized audio.
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None means do not save audio. Defaults to None.
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Returns:
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wav_base64: The base64 format of the synthesized audio.
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"""
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lang = self.config.lang
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wav_list = []
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for wav in self.executor.infer(
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text=sentence,
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lang=lang,
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am=self.config.am,
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spk_id=spk_id,
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am_block=self.am_block,
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am_pad=self.am_pad,
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voc_block=self.voc_block,
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voc_pad=self.voc_pad):
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# wav type: <class 'numpy.ndarray'> float32, convert to pcm (base64)
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wav = float2pcm(wav) # float32 to int16
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wav_bytes = wav.tobytes() # to bytes
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wav_base64 = base64.b64encode(wav_bytes).decode('utf8') # to base64
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wav_list.append(wav)
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yield wav_base64
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wav_all = np.concatenate(wav_list, axis=0)
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logger.info("The durations of audio is: {} s".format(
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len(wav_all) / self.executor.am_config.fs))
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@ -0,0 +1,100 @@
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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 os
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import time
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import requests
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from paddlespeech.server.utils.audio_process import pcm2wav
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def save_audio(buffer, audio_path) -> bool:
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if args.save_path.endswith("pcm"):
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with open(args.save_path, "wb") as f:
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f.write(buffer)
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elif args.save_path.endswith("wav"):
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with open("./tmp.pcm", "wb") as f:
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f.write(buffer)
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pcm2wav("./tmp.pcm", audio_path, channels=1, bits=16, sample_rate=24000)
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os.system("rm ./tmp.pcm")
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else:
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print("Only supports saved audio format is pcm or wav")
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return False
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return True
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def test(args):
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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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buffer = b''
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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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chunk = base64.b64decode(chunk) # bytes
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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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buffer += chunk
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final_response = time.time() - st
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duration = len(buffer) / 2.0 / 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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if args.save_path is not None:
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if save_audio(buffer, args.save_path):
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print("音频保存至：", args.save_path)
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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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parser.add_argument(
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"--save_path", type=str, help="save audio path", default=None)
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args = parser.parse_args()
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test(args)
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@ -0,0 +1,112 @@
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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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# http://www.apache.org/licenses/LICENSE-2.0
|
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#
|
||||
# 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.
|
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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',
|
||||
type=str,
|
||||
default="您好，欢迎使用语音合成服务。",
|
||||
help='A sentence to be synthesized')
|
||||
parser.add_argument('--spk_id', type=int, default=0, help='Speaker id')
|
||||
parser.add_argument('--speed', type=float, default=1.0, help='Audio speed')
|
||||
parser.add_argument(
|
||||
'--volume', type=float, default=1.0, help='Audio volume')
|
||||
parser.add_argument(
|
||||
'--sample_rate',
|
||||
type=int,
|
||||
default=0,
|
||||
help='Sampling rate, the default is the same as the model')
|
||||
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()
|
||||
test(args)
|
||||
@ -0,0 +1,126 @@
|
||||
# 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 time
|
||||
|
||||
import websocket
|
||||
|
||||
flag = 1
|
||||
st = 0.0
|
||||
all_bytes = b''
|
||||
|
||||
|
||||
class WsParam(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 on_message(ws, message):
|
||||
global flag
|
||||
global st
|
||||
global all_bytes
|
||||
|
||||
try:
|
||||
message = json.loads(message)
|
||||
audio = message["audio"]
|
||||
audio = base64.b64decode(audio) # bytes
|
||||
status = message["status"]
|
||||
all_bytes += audio
|
||||
|
||||
if status == 0:
|
||||
print("create successfully.")
|
||||
elif status == 1:
|
||||
if flag:
|
||||
print(f"首包响应：{time.time() - st} s")
|
||||
flag = 0
|
||||
elif status == 2:
|
||||
final_response = time.time() - st
|
||||
duration = len(all_bytes) / 2.0 / 24000
|
||||
print(f"尾包响应：{final_response} s")
|
||||
print(f"音频时长：{duration} s")
|
||||
print(f"RTF: {final_response / duration}")
|
||||
with open("./out.pcm", "wb") as f:
|
||||
f.write(all_bytes)
|
||||
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 = WsParam(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})
|
||||
@ -0,0 +1,160 @@
|
||||
# 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 WsParam(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 = WsParam(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()
|
||||
@ -0,0 +1,62 @@
|
||||
# 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 json
|
||||
|
||||
from fastapi import APIRouter
|
||||
from fastapi import WebSocket
|
||||
from fastapi import WebSocketDisconnect
|
||||
from starlette.websockets import WebSocketState as WebSocketState
|
||||
|
||||
from paddlespeech.cli.log import logger
|
||||
from paddlespeech.server.engine.engine_pool import get_engine_pool
|
||||
|
||||
router = APIRouter()
|
||||
|
||||
|
||||
@router.websocket('/ws/tts')
|
||||
async def websocket_endpoint(websocket: WebSocket):
|
||||
await websocket.accept()
|
||||
|
||||
try:
|
||||
# careful here, changed the source code from starlette.websockets
|
||||
assert websocket.application_state == WebSocketState.CONNECTED
|
||||
message = await websocket.receive()
|
||||
websocket._raise_on_disconnect(message)
|
||||
|
||||
# get engine
|
||||
engine_pool = get_engine_pool()
|
||||
tts_engine = engine_pool['tts']
|
||||
|
||||
# 获取 message 并转文本
|
||||
message = json.loads(message["text"])
|
||||
text_bese64 = message["text"]
|
||||
sentence = tts_engine.preprocess(text_bese64=text_bese64)
|
||||
|
||||
# run
|
||||
wav_generator = tts_engine.run(sentence)
|
||||
|
||||
while True:
|
||||
try:
|
||||
tts_results = next(wav_generator)
|
||||
resp = {"status": 1, "audio": tts_results}
|
||||
await websocket.send_json(resp)
|
||||
logger.info("streaming audio...")
|
||||
except StopIteration as e:
|
||||
resp = {"status": 2, "audio": ''}
|
||||
await websocket.send_json(resp)
|
||||
logger.info("Complete the transmission of audio streams")
|
||||
break
|
||||
|
||||
except WebSocketDisconnect:
|
||||
pass
|
||||
Loading…
Reference in new issue