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import os |
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gpt_path = os.environ.get( |
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"gpt_path", "pretrained_models/jay-e15.ckpt" |
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) |
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sovits_path = os.environ.get("sovits_path", "pretrained_models/jay_e10_s60.pth") |
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cnhubert_base_path = os.environ.get( |
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"cnhubert_base_path", "pretrained_models/chinese-hubert-base" |
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) |
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bert_path = os.environ.get( |
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"bert_path", "pretrained_models/chinese-roberta-wwm-ext-large" |
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) |
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if "_CUDA_VISIBLE_DEVICES" in os.environ: |
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os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"] |
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import gradio as gr |
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import librosa |
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import numpy as np |
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import torch |
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from transformers import AutoModelForMaskedLM, AutoTokenizer |
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from feature_extractor import cnhubert |
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cnhubert.cnhubert_base_path = cnhubert_base_path |
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from time import time as ttime |
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import datetime |
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from AR.models.t2s_lightning_module import Text2SemanticLightningModule |
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from module.mel_processing import spectrogram_torch |
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from module.models import SynthesizerTrn |
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from my_utils import load_audio |
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from text import cleaned_text_to_sequence |
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from text.cleaner import clean_text |
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device = "cuda" if torch.cuda.is_available() else "cpu" |
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is_half = eval( |
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os.environ.get("is_half", "True" if torch.cuda.is_available() else "False") |
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) |
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tokenizer = AutoTokenizer.from_pretrained(bert_path) |
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bert_model = AutoModelForMaskedLM.from_pretrained(bert_path) |
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if is_half == True: |
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bert_model = bert_model.half().to(device) |
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else: |
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bert_model = bert_model.to(device) |
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def get_bert_feature(text, word2ph): |
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with torch.no_grad(): |
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inputs = tokenizer(text, return_tensors="pt") |
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for i in inputs: |
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inputs[i] = inputs[i].to(device) |
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res = bert_model(**inputs, output_hidden_states=True) |
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res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1] |
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assert len(word2ph) == len(text) |
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phone_level_feature = [] |
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for i in range(len(word2ph)): |
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repeat_feature = res[i].repeat(word2ph[i], 1) |
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phone_level_feature.append(repeat_feature) |
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phone_level_feature = torch.cat(phone_level_feature, dim=0) |
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return phone_level_feature.T |
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n_semantic = 1024 |
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dict_s2 = torch.load(sovits_path, map_location="cpu") |
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hps = dict_s2["config"] |
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class DictToAttrRecursive: |
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def __init__(self, input_dict): |
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for key, value in input_dict.items(): |
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if isinstance(value, dict): |
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setattr(self, key, DictToAttrRecursive(value)) |
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else: |
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setattr(self, key, value) |
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hps = DictToAttrRecursive(hps) |
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hps.model.semantic_frame_rate = "25hz" |
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dict_s1 = torch.load(gpt_path, map_location="cpu") |
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config = dict_s1["config"] |
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ssl_model = cnhubert.get_model() |
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if is_half == True: |
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ssl_model = ssl_model.half().to(device) |
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else: |
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ssl_model = ssl_model.to(device) |
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vq_model = SynthesizerTrn( |
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hps.data.filter_length // 2 + 1, |
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hps.train.segment_size // hps.data.hop_length, |
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n_speakers=hps.data.n_speakers, |
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**hps.model, |
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) |
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if is_half == True: |
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vq_model = vq_model.half().to(device) |
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else: |
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vq_model = vq_model.to(device) |
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vq_model.eval() |
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print(vq_model.load_state_dict(dict_s2["weight"], strict=False)) |
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hz = 50 |
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max_sec = config["data"]["max_sec"] |
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t2s_model = Text2SemanticLightningModule(config, "ojbk", is_train=False) |
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t2s_model.load_state_dict(dict_s1["weight"]) |
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if is_half == True: |
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t2s_model = t2s_model.half() |
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t2s_model = t2s_model.to(device) |
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t2s_model.eval() |
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total = sum([param.nelement() for param in t2s_model.parameters()]) |
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print("Number of parameter: %.2fM" % (total / 1e6)) |
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def get_spepc(hps, filename): |
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audio = load_audio(filename, int(hps.data.sampling_rate)) |
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audio = torch.FloatTensor(audio) |
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audio_norm = audio |
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audio_norm = audio_norm.unsqueeze(0) |
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spec = spectrogram_torch( |
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audio_norm, |
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hps.data.filter_length, |
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hps.data.sampling_rate, |
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hps.data.hop_length, |
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hps.data.win_length, |
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center=False, |
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) |
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return spec |
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dict_language = {"Chinese": "zh", "English": "en", "Japanese": "ja"} |
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def get_tts_wav(ref_wav_path, prompt_text, prompt_language, text, text_language): |
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start_time = datetime.datetime.now() |
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print(f"---START---{start_time}---") |
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print(f"ref_wav_path: {ref_wav_path}") |
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print(f"prompt_text: {prompt_text}") |
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print(f"prompt_language: {prompt_language}") |
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print(f"text: {text}") |
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print(f"text_language: {text_language}") |
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if len(prompt_text) > 100 or len(text) > 100: |
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print("Input text is limited to 100 characters.") |
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return "Input text is limited to 100 characters.", None |
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t0 = ttime() |
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prompt_text = prompt_text.strip("\n") |
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prompt_language, text = prompt_language, text.strip("\n") |
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with torch.no_grad(): |
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wav16k, _ = librosa.load(ref_wav_path, sr=16000) |
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if len(wav16k) > 16000 * 60: |
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print("Input audio is limited to 60 seconds.") |
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return "Input audio is limited to 60 seconds.", None |
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wav16k = wav16k[: int(hps.data.sampling_rate * max_sec)] |
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wav16k = torch.from_numpy(wav16k) |
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if is_half == True: |
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wav16k = wav16k.half().to(device) |
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else: |
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wav16k = wav16k.to(device) |
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ssl_content = ssl_model.model(wav16k.unsqueeze(0))[ |
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"last_hidden_state" |
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].transpose( |
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1, 2 |
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) |
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codes = vq_model.extract_latent(ssl_content) |
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prompt_semantic = codes[0, 0] |
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t1 = ttime() |
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prompt_language = dict_language[prompt_language] |
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text_language = dict_language[text_language] |
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phones1, word2ph1, norm_text1 = clean_text(prompt_text, prompt_language) |
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phones1 = cleaned_text_to_sequence(phones1) |
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texts = text.split("\n") |
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audio_opt = [] |
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zero_wav = np.zeros( |
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int(hps.data.sampling_rate * 0.3), |
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dtype=np.float16 if is_half == True else np.float32, |
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) |
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for text in texts: |
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phones2, word2ph2, norm_text2 = clean_text(text, text_language) |
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phones2 = cleaned_text_to_sequence(phones2) |
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if prompt_language == "zh": |
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bert1 = get_bert_feature(norm_text1, word2ph1).to(device) |
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else: |
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bert1 = torch.zeros( |
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(1024, len(phones1)), |
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dtype=torch.float16 if is_half == True else torch.float32, |
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).to(device) |
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if text_language == "zh": |
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bert2 = get_bert_feature(norm_text2, word2ph2).to(device) |
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else: |
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bert2 = torch.zeros((1024, len(phones2))).to(bert1) |
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bert = torch.cat([bert1, bert2], 1) |
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all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0) |
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bert = bert.to(device).unsqueeze(0) |
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all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device) |
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prompt = prompt_semantic.unsqueeze(0).to(device) |
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t2 = ttime() |
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with torch.no_grad(): |
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pred_semantic, idx = t2s_model.model.infer_panel( |
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all_phoneme_ids, |
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all_phoneme_len, |
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prompt, |
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bert, |
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top_k=config["inference"]["top_k"], |
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early_stop_num=hz * max_sec, |
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) |
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t3 = ttime() |
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pred_semantic = pred_semantic[:, -idx:].unsqueeze( |
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0 |
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) |
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refer = get_spepc(hps, ref_wav_path) |
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if is_half == True: |
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refer = refer.half().to(device) |
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else: |
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refer = refer.to(device) |
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audio = ( |
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vq_model.decode( |
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pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer |
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) |
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.detach() |
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.cpu() |
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.numpy()[0, 0] |
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) |
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audio_opt.append(audio) |
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audio_opt.append(zero_wav) |
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t4 = ttime() |
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end_time = datetime.datetime.now() |
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dur = end_time - start_time |
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print( |
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f"Success! total time: {dur.seconds:.3f} sec,\ndetail time: {t1 - t0:.3f}, {t2 - t1:.3f}, {t3 - t2:.3f}, {t4 - t3:.3f}" |
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) |
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print(f"---END---{end_time}---") |
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return ( |
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f"Success! total time: {dur.seconds:.3f} sec,\ndetail time: {t1 - t0:.3f}, {t2 - t1:.3f}, {t3 - t2:.3f}, {t4 - t3:.3f}", |
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( |
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hps.data.sampling_rate, |
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(np.concatenate(audio_opt, 0) * 32768).astype(np.int16), |
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), |
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) |
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with gr.Blocks(title="GPT-SoVITS Zero-shot TTS Demo") as app: |
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gr.Markdown("# <center>🥳💕🎶 GPT-SoVITS 1分钟完美声音克隆!</center>") |
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gr.Markdown("## <center>🌟 支持中日英三语自动标注及训练 + 中文方言训练!Powered by [GPT-SoVITS](https://github.com/RVC-Boss/GPT-SoVITS)</center>") |
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gr.Markdown("## 请上传参考音频") |
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with gr.Row(): |
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inp_ref = gr.Audio(label="请上传数据集中的参考音频", type="filepath", value="jay_speech.wav_1361600_1516800.wav") |
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prompt_text = gr.Textbox(label="参考音频对应的文字内容", value="你喜欢这样的东西,所以你我觉得人要有一技之长呢。") |
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prompt_language = gr.Dropdown( |
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label="参考音频的语言", |
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choices=["Chinese", "English", "Japanese"], |
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value="Chinese", |
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) |
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gr.Markdown("## 开始真实拟声之旅吧!") |
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with gr.Row(): |
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text = gr.Textbox(label="需要合成的内容", lines=5) |
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text_language = gr.Dropdown( |
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label="合成内容的语言", |
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choices=["Chinese", "English", "Japanese"], |
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value="Chinese", |
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) |
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inference_button = gr.Button("开始真实拟声吧!", variant="primary") |
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with gr.Column(): |
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info = gr.Textbox(label="Info", visible=False) |
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output = gr.Audio(label="为您合成的专属音频") |
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inference_button.click( |
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get_tts_wav, |
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[inp_ref, prompt_text, prompt_language, text, text_language], |
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[info, output], |
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) |
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gr.Markdown("### <center>注意❗:请不要生成会对个人以及组织造成侵害的内容,此程序仅供科研、学习及个人娱乐使用。</center>") |
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gr.HTML(''' |
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<div class="footer"> |
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<p>🌊🏞️🎶 - 江水东流急,滔滔无尽声。 明·顾璘 |
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</p> |
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</div> |
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''') |
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app.queue(max_size=10) |
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app.launch(inbrowser=True) |
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