Merge pull request #1386 from lym0302/tts-server

[server] tts server

speechserving/speechserving/conf/tts/tts.yaml

+# 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

speechserving/speechserving/engine/tts/python/tts_engine.py

+# 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

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",

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)

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"

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