diff --git a/speechserving/speechserving/conf/tts/tts.yaml b/speechserving/speechserving/conf/tts/tts.yaml new file mode 100644 index 00000000..8e08d51c --- /dev/null +++ b/speechserving/speechserving/conf/tts/tts.yaml @@ -0,0 +1,38 @@ +# This is the parameter configuration file for TTS server. + +################################################################## +# TTS SERVER SETTING # +################################################################## +host: '0.0.0.0' +port: 8692 + +################################################################## +# ACOUSTIC MODEL SETTING # +# am choices=['speedyspeech_csmsc', 'fastspeech2_csmsc', +# 'fastspeech2_ljspeech', 'fastspeech2_aishell3', +# 'fastspeech2_vctk'] +################################################################## +am: 'fastspeech2_csmsc' +am_config: +am_ckpt: +am_stat: +phones_dict: +tones_dict: +speaker_dict: +spk_id: 0 + +################################################################## +# VOCODER SETTING # +# voc choices=['pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3', +# 'pwgan_vctk', 'mb_melgan_csmsc'] +################################################################## +voc: 'pwgan_csmsc' +voc_config: +voc_ckpt: +voc_stat: + +################################################################## +# OTHERS # +################################################################## +lang: 'zh' +device: paddle.get_device() \ No newline at end of file diff --git a/speechserving/speechserving/engine/tts/python/tts_engine.py b/speechserving/speechserving/engine/tts/python/tts_engine.py new file mode 100644 index 00000000..d790aa31 --- /dev/null +++ b/speechserving/speechserving/engine/tts/python/tts_engine.py @@ -0,0 +1,143 @@ +# Copyright (c) 2021 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 argparse +import base64 + +import librosa +import numpy as np +import soundfile as sf +import yaml +from engine.base_engine import BaseEngine + +from paddlespeech.cli.log import logger +from paddlespeech.cli.tts.infer import TTSExecutor + +__all__ = ['TTSEngine'] + + +class TTSServerExecutor(TTSExecutor): + def __init__(self): + super().__init__() + + self.parser = argparse.ArgumentParser( + prog='paddlespeech.tts', add_help=True) + self.parser.add_argument( + '--conf', + type=str, + default='./conf/tts/tts.yaml', + help='Configuration parameters.') + + +class TTSEngine(BaseEngine): + """TTS server engine + + Args: + metaclass: Defaults to Singleton. + """ + + def __init__(self, name=None): + """Initialize TTS server engine + """ + super(TTSEngine, self).__init__() + self.executor = TTSServerExecutor() + + config_path = self.executor.parser.parse_args().conf + with open(config_path, 'rt') as f: + self.conf_dict = yaml.safe_load(f) + + self.executor._init_from_path( + am=self.conf_dict["am"], + am_config=self.conf_dict["am_config"], + am_ckpt=self.conf_dict["am_ckpt"], + am_stat=self.conf_dict["am_stat"], + phones_dict=self.conf_dict["phones_dict"], + tones_dict=self.conf_dict["tones_dict"], + speaker_dict=self.conf_dict["speaker_dict"], + voc=self.conf_dict["voc"], + voc_config=self.conf_dict["voc_config"], + voc_ckpt=self.conf_dict["voc_ckpt"], + voc_stat=self.conf_dict["voc_stat"], + lang=self.conf_dict["lang"]) + + logger.info("Initialize TTS server engine successfully.") + + def postprocess(self, + wav, + original_fs: int, + target_fs: int=16000, + volume: float=1.0, + speed: float=1.0, + audio_path: str=None, + audio_format: str="wav"): + """Post-processing operations, including speech, volume, sample rate, save audio file + + Args: + wav (numpy(float)): Synthesized audio sample points + original_fs (int): original audio sample rate + target_fs (int): target audio sample rate + volume (float): target volume + speed (float): target speed + """ + + # transform sample_rate + if target_fs == 0 or target_fs > original_fs: + target_fs = original_fs + wav_tar_fs = wav + else: + wav_tar_fs = librosa.resample( + np.squeeze(wav), original_fs, target_fs) + + # transform volume + wav_vol = wav_tar_fs * volume + + # transform speed + # TODO + target_wav = wav_vol.reshape(-1, 1) + + # save audio + if audio_path is not None: + sf.write(audio_path, target_wav, target_fs) + logger.info('Wave file has been generated: {}'.format(audio_path)) + + # wav to base64 + base64_bytes = base64.b64encode(target_wav) + base64_string = base64_bytes.decode('utf-8') + wav_base64 = base64_string + + return target_fs, wav_base64 + + def run(self, + sentence: str, + spk_id: int=0, + speed: float=1.0, + volume: float=1.0, + sample_rate: int=0, + save_path: str=None, + audio_format: str="wav"): + + lang = self.conf_dict["lang"] + + self.executor.infer( + text=sentence, lang=lang, am=self.conf_dict["am"], spk_id=spk_id) + + target_sample_rate, wav_base64 = self.postprocess( + wav=self.executor._outputs['wav'].numpy(), + original_fs=self.executor.am_config.fs, + target_fs=sample_rate, + volume=volume, + speed=speed, + audio_path=save_path, + audio_format=audio_format) + + return lang, target_sample_rate, wav_base64 diff --git a/speechserving/speechserving/main.py b/speechserving/speechserving/main.py index 8374b291..d3e6edc1 100644 --- a/speechserving/speechserving/main.py +++ b/speechserving/speechserving/main.py @@ -15,11 +15,12 @@ import argparse import uvicorn import yaml -from engine.asr.python.asr_engine import ASREngine + +from engine.tts.python.tts_engine import TTSEngine from fastapi import FastAPI from restful.api import router as api_router -from utils.log import logger +from paddlespeech.cli.log import logger app = FastAPI( title="PaddleSpeech Serving API", description="Api", version="0.0.1") @@ -31,7 +32,8 @@ def init(args): app.include_router(api_router) # engine single - ASR_ENGINE = ASREngine("asr") + + TTS_ENGINE = TTSEngine() # todo others @@ -56,7 +58,8 @@ if __name__ == "__main__": "--config_file", action="store", help="yaml file of the app", - default="./conf/application.yaml") + default="./conf/tts/tts.yaml") + parser.add_argument( "--log_file", action="store", diff --git a/speechserving/speechserving/restful/api.py b/speechserving/speechserving/restful/api.py index dc309fab..e25a3158 100644 --- a/speechserving/speechserving/restful/api.py +++ b/speechserving/speechserving/restful/api.py @@ -13,9 +13,10 @@ # limitations under the License. from fastapi import APIRouter -from .asr_api import router as asr_router + from .tts_api import router as tts_router +#from .asr_api import router as asr_router router = APIRouter() -router.include_router(asr_router) +#router.include_router(asr_router) router.include_router(tts_router) diff --git a/speechserving/speechserving/restful/request.py b/speechserving/speechserving/restful/request.py index df47038f..32f157d3 100644 --- a/speechserving/speechserving/restful/request.py +++ b/speechserving/speechserving/restful/request.py @@ -16,7 +16,7 @@ from typing import Optional from pydantic import BaseModel -__all__ = ['ASRRequest, TTSRequest'] +__all__ = ['ASRRequest', 'TTSRequest'] #****************************************************************************************/ @@ -44,13 +44,25 @@ class ASRRequest(BaseModel): #************************************ TTS request ***************************************/ #****************************************************************************************/ class TTSRequest(BaseModel): - """ + """TTS request + request body example { - "audio": "exSI6ICJlbiIsCgkgICAgInBvc2l0aW9uIjogImZhbHNlIgoJf...", - "audio_format": "wav", - "sample_rate": 16000, - "lang ": "zh_cn", - "ptt ":false + "text": "你好，欢迎使用百度飞桨语音合成服务。", + "spk_id": 0, + "speed": 1.0, + "volume": 1.0, + "sample_rate": 0, + "tts_audio_path": "./tts.wav", + "audio_format": "wav" } + """ + + text: str + spk_id: int = 0 + speed: float = 1.0 + volume: float = 1.0 + sample_rate: int = 0 + save_path: str = None + audio_format: str = "wav" diff --git a/speechserving/speechserving/restful/response.py b/speechserving/speechserving/restful/response.py index de61c99c..684a37f9 100644 --- a/speechserving/speechserving/restful/response.py +++ b/speechserving/speechserving/restful/response.py @@ -16,7 +16,7 @@ from typing import Optional from pydantic import BaseModel -__all__ = ['ASRResponse'] +__all__ = ['ASRResponse', 'TTSResponse'] class Message(BaseModel): @@ -53,3 +53,36 @@ class ASRResponse(BaseModel): #****************************************************************************************/ #************************************ TTS response **************************************/ #****************************************************************************************/ +class TTSResult(BaseModel): + lang: str = "zh" + sample_rate: int + spk_id: int = 0 + speed: float = 1.0 + volume: float = 1.0 + save_path: str = None + audio: str + + +class TTSResponse(BaseModel): + """ + response example + { + "success": true, + "code": 0, + "message": { + "description": "success" + }, + "result": { + "lang": "zh", + "sample_rate": 24000, + "speed": 1.0, + "volume": 1.0, + "audio": "LTI1OTIuNjI1OTUwMzQsOTk2OS41NDk4...", + "save_path": "./tts.wav" + } + } + """ + success: bool + code: int + message: Message + result: TTSResult