# 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 ast import os from collections import OrderedDict from typing import Any from typing import List from typing import Optional from typing import Union import numpy as np import paddle import soundfile as sf import yaml from yacs.config import CfgNode from ..executor import BaseExecutor from ..log import logger from ..utils import cli_register from ..utils import download_and_decompress from ..utils import MODEL_HOME from ..utils import stats_wrapper from paddlespeech.s2t.utils.dynamic_import import dynamic_import from paddlespeech.t2s.frontend import English from paddlespeech.t2s.frontend.zh_frontend import Frontend from paddlespeech.t2s.modules.normalizer import ZScore __all__ = ['TTSExecutor'] pretrained_models = { # speedyspeech "speedyspeech_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/speedyspeech/speedyspeech_nosil_baker_ckpt_0.5.zip', 'md5': '9edce23b1a87f31b814d9477bf52afbc', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_11400.pdz', 'speech_stats': 'feats_stats.npy', 'phones_dict': 'phone_id_map.txt', 'tones_dict': 'tone_id_map.txt', }, # fastspeech2 "fastspeech2_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_baker_ckpt_0.4.zip', 'md5': '637d28a5e53aa60275612ba4393d5f22', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_76000.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', }, "fastspeech2_ljspeech-en": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_ljspeech_ckpt_0.5.zip', 'md5': 'ffed800c93deaf16ca9b3af89bfcd747', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_100000.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', }, "fastspeech2_aishell3-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_aishell3_ckpt_0.4.zip', 'md5': 'f4dd4a5f49a4552b77981f544ab3392e', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_96400.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', 'speaker_dict': 'speaker_id_map.txt', }, "fastspeech2_vctk-en": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/fastspeech2/fastspeech2_nosil_vctk_ckpt_0.5.zip', 'md5': '743e5024ca1e17a88c5c271db9779ba4', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_66200.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', 'speaker_dict': 'speaker_id_map.txt', }, # tacotron2 "tacotron2_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/tacotron2/tacotron2_csmsc_ckpt_0.2.0.zip', 'md5': '0df4b6f0bcbe0d73c5ed6df8867ab91a', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_30600.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', }, "tacotron2_ljspeech-en": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/tacotron2/tacotron2_ljspeech_ckpt_0.2.0.zip', 'md5': '6a5eddd81ae0e81d16959b97481135f3', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_60300.pdz', 'speech_stats': 'speech_stats.npy', 'phones_dict': 'phone_id_map.txt', }, # pwgan "pwgan_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/pwgan/pwg_baker_ckpt_0.4.zip', 'md5': '2e481633325b5bdf0a3823c714d2c117', 'config': 'pwg_default.yaml', 'ckpt': 'pwg_snapshot_iter_400000.pdz', 'speech_stats': 'pwg_stats.npy', }, "pwgan_ljspeech-en": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/pwgan/pwg_ljspeech_ckpt_0.5.zip', 'md5': '53610ba9708fd3008ccaf8e99dacbaf0', 'config': 'pwg_default.yaml', 'ckpt': 'pwg_snapshot_iter_400000.pdz', 'speech_stats': 'pwg_stats.npy', }, "pwgan_aishell3-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/pwgan/pwg_aishell3_ckpt_0.5.zip', 'md5': 'd7598fa41ad362d62f85ffc0f07e3d84', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_1000000.pdz', 'speech_stats': 'feats_stats.npy', }, "pwgan_vctk-en": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/pwgan/pwg_vctk_ckpt_0.1.1.zip', 'md5': 'b3da1defcde3e578be71eb284cb89f2c', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_1500000.pdz', 'speech_stats': 'feats_stats.npy', }, # mb_melgan "mb_melgan_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/mb_melgan/mb_melgan_csmsc_ckpt_0.1.1.zip', 'md5': 'ee5f0604e20091f0d495b6ec4618b90d', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_1000000.pdz', 'speech_stats': 'feats_stats.npy', }, # style_melgan "style_melgan_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/style_melgan/style_melgan_csmsc_ckpt_0.1.1.zip', 'md5': '5de2d5348f396de0c966926b8c462755', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_1500000.pdz', 'speech_stats': 'feats_stats.npy', }, # hifigan "hifigan_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/hifigan/hifigan_csmsc_ckpt_0.1.1.zip', 'md5': 'dd40a3d88dfcf64513fba2f0f961ada6', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_2500000.pdz', 'speech_stats': 'feats_stats.npy', }, # wavernn "wavernn_csmsc-zh": { 'url': 'https://paddlespeech.bj.bcebos.com/Parakeet/released_models/wavernn/wavernn_csmsc_ckpt_0.2.0.zip', 'md5': 'ee37b752f09bcba8f2af3b777ca38e13', 'config': 'default.yaml', 'ckpt': 'snapshot_iter_400000.pdz', 'speech_stats': 'feats_stats.npy', } } model_alias = { # acoustic model "speedyspeech": "paddlespeech.t2s.models.speedyspeech:SpeedySpeech", "speedyspeech_inference": "paddlespeech.t2s.models.speedyspeech:SpeedySpeechInference", "fastspeech2": "paddlespeech.t2s.models.fastspeech2:FastSpeech2", "fastspeech2_inference": "paddlespeech.t2s.models.fastspeech2:FastSpeech2Inference", "tacotron2": "paddlespeech.t2s.models.tacotron2:Tacotron2", "tacotron2_inference": "paddlespeech.t2s.models.tacotron2:Tacotron2Inference", # voc "pwgan": "paddlespeech.t2s.models.parallel_wavegan:PWGGenerator", "pwgan_inference": "paddlespeech.t2s.models.parallel_wavegan:PWGInference", "mb_melgan": "paddlespeech.t2s.models.melgan:MelGANGenerator", "mb_melgan_inference": "paddlespeech.t2s.models.melgan:MelGANInference", "style_melgan": "paddlespeech.t2s.models.melgan:StyleMelGANGenerator", "style_melgan_inference": "paddlespeech.t2s.models.melgan:StyleMelGANInference", "hifigan": "paddlespeech.t2s.models.hifigan:HiFiGANGenerator", "hifigan_inference": "paddlespeech.t2s.models.hifigan:HiFiGANInference", "wavernn": "paddlespeech.t2s.models.wavernn:WaveRNN", "wavernn_inference": "paddlespeech.t2s.models.wavernn:WaveRNNInference", } @cli_register( name='paddlespeech.tts', description='Text to Speech infer command.') class TTSExecutor(BaseExecutor): def __init__(self): super().__init__() self.parser = argparse.ArgumentParser( prog='paddlespeech.tts', add_help=True) self.parser.add_argument( '--input', type=str, default=None, help='Input text to generate.') # acoustic model self.parser.add_argument( '--am', type=str, default='fastspeech2_csmsc', choices=[ 'speedyspeech_csmsc', 'fastspeech2_csmsc', 'fastspeech2_ljspeech', 'fastspeech2_aishell3', 'fastspeech2_vctk', 'tacotron2_csmsc', 'tacotron2_ljspeech', ], help='Choose acoustic model type of tts task.') self.parser.add_argument( '--am_config', type=str, default=None, help='Config of acoustic model. Use deault config when it is None.') self.parser.add_argument( '--am_ckpt', type=str, default=None, help='Checkpoint file of acoustic model.') self.parser.add_argument( "--am_stat", type=str, default=None, help="mean and standard deviation used to normalize spectrogram when training acoustic model." ) self.parser.add_argument( "--phones_dict", type=str, default=None, help="phone vocabulary file.") self.parser.add_argument( "--tones_dict", type=str, default=None, help="tone vocabulary file.") self.parser.add_argument( "--speaker_dict", type=str, default=None, help="speaker id map file.") self.parser.add_argument( '--spk_id', type=int, default=0, help='spk id for multi speaker acoustic model') # vocoder self.parser.add_argument( '--voc', type=str, default='pwgan_csmsc', choices=[ 'pwgan_csmsc', 'pwgan_ljspeech', 'pwgan_aishell3', 'pwgan_vctk', 'mb_melgan_csmsc', 'style_melgan_csmsc', 'hifigan_csmsc', 'wavernn_csmsc', ], help='Choose vocoder type of tts task.') self.parser.add_argument( '--voc_config', type=str, default=None, help='Config of voc. Use deault config when it is None.') self.parser.add_argument( '--voc_ckpt', type=str, default=None, help='Checkpoint file of voc.') self.parser.add_argument( "--voc_stat", type=str, default=None, help="mean and standard deviation used to normalize spectrogram when training voc." ) # other self.parser.add_argument( '--lang', type=str, default='zh', help='Choose model language. zh or en') self.parser.add_argument( '--device', type=str, default=paddle.get_device(), help='Choose device to execute model inference.') self.parser.add_argument( '--output', type=str, default='output.wav', help='output file name') self.parser.add_argument( '--job_dump_result', type=ast.literal_eval, default=False, help='Save job result into file.') def _get_pretrained_path(self, tag: str) -> os.PathLike: """ Download and returns pretrained resources path of current task. """ support_models = list(pretrained_models.keys()) assert tag in pretrained_models, 'The model "{}" you want to use has not been supported, please choose other models.\nThe support models includes:\n\t\t{}\n'.format( tag, '\n\t\t'.join(support_models)) res_path = os.path.join(MODEL_HOME, tag) decompressed_path = download_and_decompress(pretrained_models[tag], res_path) decompressed_path = os.path.abspath(decompressed_path) logger.info( 'Use pretrained model stored in: {}'.format(decompressed_path)) return decompressed_path def _init_from_path( self, am: str='fastspeech2_csmsc', am_config: Optional[os.PathLike]=None, am_ckpt: Optional[os.PathLike]=None, am_stat: Optional[os.PathLike]=None, phones_dict: Optional[os.PathLike]=None, tones_dict: Optional[os.PathLike]=None, speaker_dict: Optional[os.PathLike]=None, voc: str='pwgan_csmsc', voc_config: Optional[os.PathLike]=None, voc_ckpt: Optional[os.PathLike]=None, voc_stat: Optional[os.PathLike]=None, lang: str='zh', ): """ Init model and other resources from a specific path. """ if hasattr(self, 'am_inference') and hasattr(self, 'voc_inference'): logger.info('Models had been initialized.') return # am am_tag = am + '-' + lang if am_ckpt is None or am_config is None or am_stat is None or phones_dict is None: am_res_path = self._get_pretrained_path(am_tag) self.am_res_path = am_res_path self.am_config = os.path.join(am_res_path, pretrained_models[am_tag]['config']) self.am_ckpt = os.path.join(am_res_path, pretrained_models[am_tag]['ckpt']) self.am_stat = os.path.join( am_res_path, pretrained_models[am_tag]['speech_stats']) # must have phones_dict in acoustic self.phones_dict = os.path.join( am_res_path, pretrained_models[am_tag]['phones_dict']) print("self.phones_dict:", self.phones_dict) logger.info(am_res_path) logger.info(self.am_config) logger.info(self.am_ckpt) else: self.am_config = os.path.abspath(am_config) self.am_ckpt = os.path.abspath(am_ckpt) self.am_stat = os.path.abspath(am_stat) self.phones_dict = os.path.abspath(phones_dict) self.am_res_path = os.path.dirname(os.path.abspath(self.am_config)) print("self.phones_dict:", self.phones_dict) # for speedyspeech self.tones_dict = None if 'tones_dict' in pretrained_models[am_tag]: self.tones_dict = os.path.join( am_res_path, pretrained_models[am_tag]['tones_dict']) if tones_dict: self.tones_dict = tones_dict # for multi speaker fastspeech2 self.speaker_dict = None if 'speaker_dict' in pretrained_models[am_tag]: self.speaker_dict = os.path.join( am_res_path, pretrained_models[am_tag]['speaker_dict']) if speaker_dict: self.speaker_dict = speaker_dict # voc voc_tag = voc + '-' + lang if voc_ckpt is None or voc_config is None or voc_stat is None: voc_res_path = self._get_pretrained_path(voc_tag) self.voc_res_path = voc_res_path self.voc_config = os.path.join(voc_res_path, pretrained_models[voc_tag]['config']) self.voc_ckpt = os.path.join(voc_res_path, pretrained_models[voc_tag]['ckpt']) self.voc_stat = os.path.join( voc_res_path, pretrained_models[voc_tag]['speech_stats']) logger.info(voc_res_path) logger.info(self.voc_config) logger.info(self.voc_ckpt) else: self.voc_config = os.path.abspath(voc_config) self.voc_ckpt = os.path.abspath(voc_ckpt) self.voc_stat = os.path.abspath(voc_stat) self.voc_res_path = os.path.dirname( os.path.abspath(self.voc_config)) # Init body. with open(self.am_config) as f: self.am_config = CfgNode(yaml.safe_load(f)) with open(self.voc_config) as f: self.voc_config = CfgNode(yaml.safe_load(f)) with open(self.phones_dict, "r") as f: phn_id = [line.strip().split() for line in f.readlines()] vocab_size = len(phn_id) print("vocab_size:", vocab_size) tone_size = None if self.tones_dict: with open(self.tones_dict, "r") as f: tone_id = [line.strip().split() for line in f.readlines()] tone_size = len(tone_id) print("tone_size:", tone_size) spk_num = None if self.speaker_dict: with open(self.speaker_dict, 'rt') as f: spk_id = [line.strip().split() for line in f.readlines()] spk_num = len(spk_id) print("spk_num:", spk_num) # frontend if lang == 'zh': self.frontend = Frontend( phone_vocab_path=self.phones_dict, tone_vocab_path=self.tones_dict) elif lang == 'en': self.frontend = English(phone_vocab_path=self.phones_dict) print("frontend done!") # acoustic model odim = self.am_config.n_mels # model: {model_name}_{dataset} am_name = am[:am.rindex('_')] am_class = dynamic_import(am_name, model_alias) am_inference_class = dynamic_import(am_name + '_inference', model_alias) if am_name == 'fastspeech2': am = am_class( idim=vocab_size, odim=odim, spk_num=spk_num, **self.am_config["model"]) elif am_name == 'speedyspeech': am = am_class( vocab_size=vocab_size, tone_size=tone_size, **self.am_config["model"]) elif am_name == 'tacotron2': am = am_class(idim=vocab_size, odim=odim, **self.am_config["model"]) am.set_state_dict(paddle.load(self.am_ckpt)["main_params"]) am.eval() am_mu, am_std = np.load(self.am_stat) am_mu = paddle.to_tensor(am_mu) am_std = paddle.to_tensor(am_std) am_normalizer = ZScore(am_mu, am_std) self.am_inference = am_inference_class(am_normalizer, am) self.am_inference.eval() print("acoustic model done!") # vocoder # model: {model_name}_{dataset} voc_name = voc[:voc.rindex('_')] voc_class = dynamic_import(voc_name, model_alias) voc_inference_class = dynamic_import(voc_name + '_inference', model_alias) if voc_name != 'wavernn': voc = voc_class(**self.voc_config["generator_params"]) voc.set_state_dict(paddle.load(self.voc_ckpt)["generator_params"]) voc.remove_weight_norm() voc.eval() else: voc = voc_class(**self.voc_config["model"]) voc.set_state_dict(paddle.load(self.voc_ckpt)["main_params"]) voc.eval() voc_mu, voc_std = np.load(self.voc_stat) voc_mu = paddle.to_tensor(voc_mu) voc_std = paddle.to_tensor(voc_std) voc_normalizer = ZScore(voc_mu, voc_std) self.voc_inference = voc_inference_class(voc_normalizer, voc) self.voc_inference.eval() print("voc done!") def preprocess(self, input: Any, *args, **kwargs): """ Input preprocess and return paddle.Tensor stored in self._inputs. Input content can be a text(tts), a file(asr, cls), a stream(not supported yet) or anything needed. Args: input (Any): Input text/file/stream or other content. """ pass @paddle.no_grad() def infer(self, text: str, lang: str='zh', am: str='fastspeech2_csmsc', spk_id: int=0): """ Model inference and result stored in self.output. """ am_name = am[:am.rindex('_')] am_dataset = am[am.rindex('_') + 1:] get_tone_ids = False merge_sentences = False if am_name == 'speedyspeech': get_tone_ids = True if lang == 'zh': input_ids = self.frontend.get_input_ids( text, merge_sentences=merge_sentences, get_tone_ids=get_tone_ids) phone_ids = input_ids["phone_ids"] if get_tone_ids: tone_ids = input_ids["tone_ids"] elif lang == 'en': input_ids = self.frontend.get_input_ids( text, merge_sentences=merge_sentences) phone_ids = input_ids["phone_ids"] else: print("lang should in {'zh', 'en'}!") flags = 0 for i in range(len(phone_ids)): part_phone_ids = phone_ids[i] # am if am_name == 'speedyspeech': part_tone_ids = tone_ids[i] mel = self.am_inference(part_phone_ids, part_tone_ids) # fastspeech2 else: # multi speaker if am_dataset in {"aishell3", "vctk"}: mel = self.am_inference( part_phone_ids, spk_id=paddle.to_tensor(spk_id)) else: mel = self.am_inference(part_phone_ids) # voc wav = self.voc_inference(mel) if flags == 0: wav_all = wav flags = 1 else: wav_all = paddle.concat([wav_all, wav]) self._outputs['wav'] = wav_all def postprocess(self, output: str='output.wav') -> Union[str, os.PathLike]: """ Output postprocess and return results. This method get model output from self._outputs and convert it into human-readable results. Returns: Union[str, os.PathLike]: Human-readable results such as texts and audio files. """ output = os.path.abspath(os.path.expanduser(output)) sf.write( output, self._outputs['wav'].numpy(), samplerate=self.am_config.fs) return output def execute(self, argv: List[str]) -> bool: """ Command line entry. """ args = self.parser.parse_args(argv) am = args.am am_config = args.am_config am_ckpt = args.am_ckpt am_stat = args.am_stat phones_dict = args.phones_dict print("phones_dict:", phones_dict) tones_dict = args.tones_dict speaker_dict = args.speaker_dict voc = args.voc voc_config = args.voc_config voc_ckpt = args.voc_ckpt voc_stat = args.voc_stat lang = args.lang device = args.device spk_id = args.spk_id job_dump_result = args.job_dump_result task_source = self.get_task_source(args.input) task_results = OrderedDict() has_exceptions = False for id_, input_ in task_source.items(): if len(task_source) > 1: assert isinstance(args.output, str) and args.output.endswith('.wav') output = args.output.replace('.wav', f'_{id_}.wav') else: output = args.output try: res = self( text=input_, # acoustic model related am=am, am_config=am_config, am_ckpt=am_ckpt, am_stat=am_stat, phones_dict=phones_dict, tones_dict=tones_dict, speaker_dict=speaker_dict, spk_id=spk_id, # vocoder related voc=voc, voc_config=voc_config, voc_ckpt=voc_ckpt, voc_stat=voc_stat, # other lang=lang, device=device, output=output) task_results[id_] = res except Exception as e: has_exceptions = True task_results[id_] = f'{e.__class__.__name__}: {e}' self.process_task_results(args.input, task_results, job_dump_result) if has_exceptions: return False else: return True @stats_wrapper def __call__(self, text: str, am: str='fastspeech2_csmsc', am_config: Optional[os.PathLike]=None, am_ckpt: Optional[os.PathLike]=None, am_stat: Optional[os.PathLike]=None, spk_id: int=0, phones_dict: Optional[os.PathLike]=None, tones_dict: Optional[os.PathLike]=None, speaker_dict: Optional[os.PathLike]=None, voc: str='pwgan_csmsc', voc_config: Optional[os.PathLike]=None, voc_ckpt: Optional[os.PathLike]=None, voc_stat: Optional[os.PathLike]=None, lang: str='zh', device: str=paddle.get_device(), output: str='output.wav'): """ Python API to call an executor. """ paddle.set_device(device) self._init_from_path( am=am, am_config=am_config, am_ckpt=am_ckpt, am_stat=am_stat, phones_dict=phones_dict, tones_dict=tones_dict, speaker_dict=speaker_dict, voc=voc, voc_config=voc_config, voc_ckpt=voc_ckpt, voc_stat=voc_stat, lang=lang) self.infer(text=text, lang=lang, am=am, spk_id=spk_id) res = self.postprocess(output=output) return res