from cached_path import cached_path # from dp.phonemizer import Phonemizer print("NLTK") import nltk nltk.download('punkt') print("SCIPY") from scipy.io.wavfile import write print("TORCH STUFF") import torch print("START") torch.manual_seed(0) torch.backends.cudnn.benchmark = False torch.backends.cudnn.deterministic = True import random random.seed(0) import numpy as np np.random.seed(0) # load packages import time import random import yaml from munch import Munch import numpy as np import torch from torch import nn import torch.nn.functional as F import torchaudio import librosa from nltk.tokenize import word_tokenize from models import * from utils import * from text_utils import TextCleaner textclenaer = TextCleaner() to_mel = torchaudio.transforms.MelSpectrogram( n_mels=80, n_fft=2048, win_length=1200, hop_length=300) mean, std = -4, 4 def length_to_mask(lengths): mask = torch.arange(lengths.max()).unsqueeze(0).expand(lengths.shape[0], -1).type_as(lengths) mask = torch.gt(mask+1, lengths.unsqueeze(1)) return mask def preprocess(wave): wave_tensor = torch.from_numpy(wave).float() mel_tensor = to_mel(wave_tensor) mel_tensor = (torch.log(1e-5 + mel_tensor.unsqueeze(0)) - mean) / std return mel_tensor def compute_style(path): wave, sr = librosa.load(path, sr=24000) audio, index = librosa.effects.trim(wave, top_db=30) if sr != 24000: audio = librosa.resample(audio, sr, 24000) mel_tensor = preprocess(audio).to(device) with torch.no_grad(): ref_s = model.style_encoder(mel_tensor.unsqueeze(1)) ref_p = model.predictor_encoder(mel_tensor.unsqueeze(1)) return torch.cat([ref_s, ref_p], dim=1) device = 'cpu' if torch.cuda.is_available(): device = 'cuda' elif torch.backends.mps.is_available(): print("MPS would be available but cannot be used rn") # device = 'mps' # config = yaml.safe_load(open("Models/LibriTTS/config.yml")) config = yaml.safe_load(open(str(cached_path("hf://yl4579/StyleTTS2-LibriTTS/Models/LibriTTS/config.yml")))) # load pretrained ASR model ASR_config = config.get('ASR_config', False) ASR_path = config.get('ASR_path', False) text_aligner = load_ASR_models(ASR_path, ASR_config) # load pretrained F0 model F0_path = config.get('F0_path', False) pitch_extractor = load_F0_models(F0_path) # load BERT model from Utils.PLBERT.util import load_plbert BERT_path = config.get('PLBERT_dir', False) plbert = load_plbert(BERT_path) model_params = recursive_munch(config['model_params']) model = build_model(model_params, text_aligner, pitch_extractor, plbert) _ = [model[key].eval() for key in model] _ = [model[key].to(device) for key in model] # params_whole = torch.load("Models/LibriTTS/epochs_2nd_00020.pth", map_location='cpu') params_whole = torch.load(str(cached_path("hf://yl4579/StyleTTS2-LibriTTS/Models/LibriTTS/epochs_2nd_00020.pth")), map_location='cpu') params = params_whole['net'] for key in model: if key in params: print('%s loaded' % key) try: model[key].load_state_dict(params[key]) except: from collections import OrderedDict state_dict = params[key] new_state_dict = OrderedDict() for k, v in state_dict.items(): name = k[7:] # remove `module.` new_state_dict[name] = v # load params model[key].load_state_dict(new_state_dict, strict=False) # except: # _load(params[key], model[key]) _ = [model[key].eval() for key in model] from Modules.diffusion.sampler import DiffusionSampler, ADPM2Sampler, KarrasSchedule sampler = DiffusionSampler( model.diffusion.diffusion, sampler=ADPM2Sampler(), sigma_schedule=KarrasSchedule(sigma_min=0.0001, sigma_max=3.0, rho=9.0), # empirical parameters clamp=False ) voicelist = ['f-us-1', 'f-us-2', 'f-us-3', 'f-us-4', 'm-us-1', 'm-us-2', 'm-us-3', 'm-us-4'] voices = {} # todo: cache computed style, load using pickle for v in voicelist: print(f"Loading voice {v}") voices[v] = compute_style(f'voices/{v}.wav') import pickle with open('voices.pkl', 'wb') as f: pickle.dump(voices, f)