import os def list_files_tree(directory, indent=""): items = os.listdir(directory) for i, item in enumerate(items): prefix = "└── " if i == len(items) - 1 else "├── " print(indent + prefix + item) item_path = os.path.join(directory, item) if os.path.isdir(item_path): next_indent = indent + (" " if i == len(items) - 1 else "│ ") list_files_tree(item_path, next_indent) from huggingface_hub import snapshot_download print("Models...") models_id = """None1145/GPT-SoVITS-Lappland-the-Decadenza None1145/GPT-SoVITS-Lappland None1145/GPT-SoVITS-Theresa None1145/GPT-SoVITS-Vulpisfoglia None1145/GPT-SoVITS-Rosmontis None1145/GPT-SoVITS-Theresa-Recording""" for model_id in models_id.split("\n"): if model_id in ["", " "]: break print(f"{model_id}...") snapshot_download(repo_id=model_id, local_dir=f"./Models/{model_id}") print(f"{model_id}!!!") print("Models!!!") print("PretrainedModels...") base_model_id = "None1145/GPT-SoVITS-Base" snapshot_download(repo_id=base_model_id, local_dir=f"./PretrainedModels/{base_model_id}") print("PretrainedModels!!!") list_files_tree("./") cnhubert_base_path = f"./PretrainedModels/{base_model_id}/chinese-hubert-base" bert_path = f"./PretrainedModels/{base_model_id}/chinese-roberta-wwm-ext-large" import gradio as gr from transformers import AutoModelForMaskedLM, AutoTokenizer import sys, torch, numpy as np from pathlib import Path from pydub import AudioSegment import librosa, math, traceback, requests, argparse, torch, multiprocessing, pandas as pd, torch.multiprocessing as mp, soundfile from random import shuffle from AR.utils import get_newest_ckpt from glob import glob from tqdm import tqdm from feature_extractor import cnhubert cnhubert.cnhubert_base_path=cnhubert_base_path from io import BytesIO from module.models import SynthesizerTrn from AR.models.t2s_lightning_module import Text2SemanticLightningModule from AR.utils.io import load_yaml_config from text import cleaned_text_to_sequence from text.cleaner import text_to_sequence, clean_text from time import time as ttime from module.mel_processing import spectrogram_torch from my_utils import load_audio import re import logging logging.getLogger('httpx').setLevel(logging.WARNING) logging.getLogger('httpcore').setLevel(logging.WARNING) logging.getLogger('multipart').setLevel(logging.WARNING) device = "cpu" is_half = False tokenizer = AutoTokenizer.from_pretrained(bert_path) bert_model=AutoModelForMaskedLM.from_pretrained(bert_path) if(is_half==True):bert_model=bert_model.half().to(device) else:bert_model=bert_model.to(device) def get_bert_feature(text, word2ph): with torch.no_grad(): inputs = tokenizer(text, return_tensors="pt") for i in inputs: inputs[i] = inputs[i].to(device) res = bert_model(**inputs, output_hidden_states=True) res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1] assert len(word2ph) == len(text) phone_level_feature = [] for i in range(len(word2ph)): repeat_feature = res[i].repeat(word2ph[i], 1) phone_level_feature.append(repeat_feature) phone_level_feature = torch.cat(phone_level_feature, dim=0) return phone_level_feature.T def load_model(sovits_path, gpt_path): n_semantic = 1024 dict_s2 = torch.load(sovits_path, map_location="cpu") hps = dict_s2["config"] class DictToAttrRecursive: def __init__(self, input_dict): for key, value in input_dict.items(): if isinstance(value, dict): setattr(self, key, DictToAttrRecursive(value)) else: setattr(self, key, value) hps = DictToAttrRecursive(hps) hps.model.semantic_frame_rate = "25hz" dict_s1 = torch.load(gpt_path, map_location="cpu") config = dict_s1["config"] ssl_model = cnhubert.get_model() if (is_half == True): ssl_model = ssl_model.half().to(device) else: ssl_model = ssl_model.to(device) vq_model = SynthesizerTrn( hps.data.filter_length // 2 + 1, hps.train.segment_size // hps.data.hop_length, n_speakers=hps.data.n_speakers, **hps.model) if (is_half == True): vq_model = vq_model.half().to(device) else: vq_model = vq_model.to(device) vq_model.eval() vq_model.load_state_dict(dict_s2["weight"], strict=False) hz = 50 max_sec = config['data']['max_sec'] t2s_model = Text2SemanticLightningModule(config, "ojbk", is_train=False) t2s_model.load_state_dict(dict_s1["weight"]) if (is_half == True): t2s_model = t2s_model.half() t2s_model = t2s_model.to(device) t2s_model.eval() total = sum([param.nelement() for param in t2s_model.parameters()]) print("Number of parameter: %.2fM" % (total / 1e6)) return vq_model, ssl_model, t2s_model, hps, config, hz, max_sec def get_spepc(hps, filename): audio=load_audio(filename,int(hps.data.sampling_rate)) audio=torch.FloatTensor(audio) audio_norm = audio audio_norm = audio_norm.unsqueeze(0) spec = spectrogram_torch(audio_norm, hps.data.filter_length,hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,center=False) return spec def create_tts_fn(vq_model, ssl_model, t2s_model, hps, config, hz, max_sec): def tts_fn(ref_wav_path, prompt_text, prompt_language, text, text_language): t0 = ttime() prompt_text=prompt_text.strip("\n") prompt_language,text=prompt_language,text.strip("\n") print(text) # if len(text) > 50: # return f"Error: Text is too long, ({len(text)}>50)", None with torch.no_grad(): wav16k, sr = librosa.load(ref_wav_path, sr=16000) wav16k = torch.from_numpy(wav16k) if(is_half==True):wav16k=wav16k.half().to(device) else:wav16k=wav16k.to(device) ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) codes = vq_model.extract_latent(ssl_content) prompt_semantic = codes[0, 0] t1 = ttime() phones1, word2ph1, norm_text1 = clean_text(prompt_text, prompt_language) phones1=cleaned_text_to_sequence(phones1) texts=text.split("\n") audio_opt = [] zero_wav=np.zeros(int(hps.data.sampling_rate*0.3),dtype=np.float16 if is_half==True else np.float32) for text in texts: phones2, word2ph2, norm_text2 = clean_text(text, text_language) phones2 = cleaned_text_to_sequence(phones2) if(prompt_language=="zh"):bert1 = get_bert_feature(norm_text1, word2ph1).to(device) else:bert1 = torch.zeros((1024, len(phones1)),dtype=torch.float16 if is_half==True else torch.float32).to(device) if(text_language=="zh"):bert2 = get_bert_feature(norm_text2, word2ph2).to(device) else:bert2 = torch.zeros((1024, len(phones2))).to(bert1) bert = torch.cat([bert1, bert2], 1) all_phoneme_ids = torch.LongTensor(phones1+phones2).to(device).unsqueeze(0) bert = bert.to(device).unsqueeze(0) all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device) prompt = prompt_semantic.unsqueeze(0).to(device) t2 = ttime() with torch.no_grad(): pred_semantic,idx = t2s_model.model.infer_panel( all_phoneme_ids, all_phoneme_len, prompt, bert, top_k=config['inference']['top_k'], early_stop_num=hz * max_sec) t3 = ttime() pred_semantic = pred_semantic[:,-idx:].unsqueeze(0) refer = get_spepc(hps, ref_wav_path)#.to(device) if(is_half==True):refer=refer.half().to(device) else:refer=refer.to(device) audio = vq_model.decode(pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer).detach().cpu().numpy()[0, 0] audio_opt.append(audio) audio_opt.append(zero_wav) t4 = ttime() print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3)) return "Success", (hps.data.sampling_rate,(np.concatenate(audio_opt,0)*32768).astype(np.int16)) return tts_fn splits={",","。","?","!",",",".","?","!","~",":",":","—","…",} def split(todo_text): todo_text = todo_text.replace("……", "。").replace("——", ",") if (todo_text[-1] not in splits): todo_text += "。" i_split_head = i_split_tail = 0 len_text = len(todo_text) todo_texts = [] while (1): if (i_split_head >= len_text): break if (todo_text[i_split_head] in splits): i_split_head += 1 todo_texts.append(todo_text[i_split_tail:i_split_head]) i_split_tail = i_split_head else: i_split_head += 1 return todo_texts def change_reference_audio(prompt_text, transcripts): return transcripts[prompt_text] def get_audio_duration(path): audio = AudioSegment.from_wav(path) return len(audio) / 1000 def select_audio_file(wav_paths): import random eligible_files = [path for path in wav_paths if 2 <= get_audio_duration(path) <= 5] if eligible_files: selected_file = random.choice(eligible_files) else: selected_file = random.choice(wav_paths) return selected_file models = [] models_info = {} models_folder_path = "./Models/None1145" folder_names = [name for name in os.listdir(models_folder_path) if os.path.isdir(os.path.join(models_folder_path, name))] for folder_name in folder_names: speaker = folder_name[11:] models_info[speaker] = {} models_info[speaker]["title"] = speaker pattern = re.compile(r"s(\d+)\.pth$") max_value = -1 max_file = None sovits_path = f"{models_folder_path}/{folder_name}/SoVITS_weights" for filename in os.listdir(sovits_path): match = pattern.search(filename) if match: value = int(match.group(1)) if value > max_value: max_value = value max_file = filename models_info[speaker]["sovits_weight"] = f"{sovits_path}/{max_file}" pattern = re.compile(r"e(\d+)\.ckpt$") max_value = -1 max_file = None gpt_path = f"{models_folder_path}/{folder_name}/GPT_weights" try: for filename in os.listdir(gpt_path): match = pattern.search(filename) if match: value = int(match.group(1)) if value > max_value: max_value = value max_file = filename models_info[speaker]["gpt_weight"] = f"{gpt_path}/{max_file}" except: models_info[speaker]["gpt_weight"] = f"{base_model_id}/{GPT.ckpt}" data_path = f"{models_folder_path}/{folder_name}/Data" models_info[speaker]["transcript"] = {} wavs = [] tmp = {} with open(f"{data_path}/{speaker}.list", "r", encoding="utf-8") as f: for line in f.read().split("\n"): try: wav = f"{models_folder_path}/{folder_name}/Data/{line.split('|')[0].split('/')[1]}" except: break text = line.split("|")[3] print(wav, text) wavs.append(wav) tmp[wav] = text models_info[speaker]["transcript"][text] = wav models_info[speaker]["example_reference"] = tmp[select_audio_file(wavs)] print(models_info) for speaker in models_info: speaker_info = models_info[speaker] title = speaker_info["title"] sovits_weight = speaker_info["sovits_weight"] gpt_weight = speaker_info["gpt_weight"] model_id = "None1145/GPT-SoVITS-Base" vq_model, ssl_model, t2s_model, hps, config, hz, max_sec = load_model(sovits_weight, gpt_weight) # vq_model, ssl_model, t2s_model, hps, config, hz, max_sec = load_model(sovits_weight, f"./PretrainedModels/{model_id}/GPT.ckpt") models.append( ( speaker, title, speaker_info["transcript"], speaker_info["example_reference"], create_tts_fn( vq_model, ssl_model, t2s_model, hps, config, hz, max_sec ) ) ) print(models) with gr.Blocks() as app: with gr.Tabs(): for (name, title, transcript, example_reference, tts_fn) in models: with gr.TabItem(name): with gr.Row(): gr.Markdown( '