import os, re, logging import LangSegment import pdb import torch import gradio as gr from transformers import AutoModelForMaskedLM, AutoTokenizer import numpy as np import librosa from feature_extractor import cnhubert from module.models import SynthesizerTrn from AR.models.t2s_lightning_module import Text2SemanticLightningModule from text import cleaned_text_to_sequence from text.cleaner import clean_text from time import time as ttime from module.mel_processing import spectrogram_torch from tools.my_utils import load_audio from tools.i18n.i18n import I18nAuto import scipy.io.wavfile as wavfile device = "cuda" if torch.cuda.is_available() else "cpu" i18n = I18nAuto() dict_language = { i18n("中文"): "all_zh", # 全部按中文识别 i18n("英文"): "en", # 全部按英文识别#######不变 i18n("日文"): "all_ja", # 全部按日文识别 i18n("中英混合"): "zh", # 按中英混合识别####不变 i18n("日英混合"): "ja", # 按日英混合识别####不变 i18n("多语种混合"): "auto", # 多语种启动切分识别语种 } is_share = os.environ.get("is_share", "False") is_share = eval(is_share) if "_CUDA_VISIBLE_DEVICES" in os.environ: os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"] half_precision = True is_half = half_precision and torch.cuda.is_available() splits = {",", "。", "?", "!", ",", ".", "?", "!", "~", ":", ":", "—", "…", } punctuation = set(['!', '?', '…', ',', '.', '-', " "]) class DictToAttrRecursive(dict): def __init__(self, input_dict): super().__init__(input_dict) for key, value in input_dict.items(): if isinstance(value, dict): value = DictToAttrRecursive(value) self[key] = value setattr(self, key, value) def __getattr__(self, item): try: return self[item] except KeyError: raise AttributeError(f"Attribute {item} not found") def __setattr__(self, key, value): if isinstance(value, dict): value = DictToAttrRecursive(value) super(DictToAttrRecursive, self).__setitem__(key, value) super().__setattr__(key, value) def __delattr__(self, item): try: del self[item] except KeyError: raise AttributeError(f"Attribute {item} not found") def replace_consecutive_punctuation(text): punctuations = ''.join(re.escape(p) for p in punctuation) pattern = f'([{punctuations}])([{punctuations}])+' result = re.sub(pattern, r'\1', text) return result def get_first(text): pattern = "[" + "".join(re.escape(sep) for sep in splits) + "]" text = re.split(pattern, text)[0].strip() return text 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 # 结尾一定有标点,所以直接跳出即可,最后一段在上次已加入 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 # 四句一切 def cut1(inp): inp = inp.strip("\n") inps = split(inp) split_idx = list(range(0, len(inps), 4)) split_idx[-1] = None if len(split_idx) > 1: opts = [] for idx in range(len(split_idx) - 1): opts.append("".join(inps[split_idx[idx]: split_idx[idx + 1]])) else: opts = [inp] opts = [item for item in opts if not set(item).issubset(punctuation)] return "\n".join(opts) # 句号切 def cut3(inp): inp = inp.strip("\n") opts = ["%s" % item for item in inp.strip("。").split("。")] opts = [item for item in opts if not set(item).issubset(punctuation)] return "\n".join(opts) def process_text(texts): _text = [] if all(text in [None, " ", "\n", ""] for text in texts): raise ValueError(i18n("请输入有效文本")) for text in texts: if text in [None, " ", ""]: pass else: _text.append(text) return _text def merge_short_text_in_array(texts, threshold): if (len(texts)) < 2: return texts result = [] text = "" for ele in texts: text += ele if len(text) >= threshold: result.append(text) text = "" if (len(text) > 0): if len(result) == 0: result.append(text) else: result[len(result) - 1] += text return result def clean_text_inf(text, language): phones, word2ph, norm_text = clean_text(text, language) phones = cleaned_text_to_sequence(phones) return phones, word2ph, norm_text 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 dtype = torch.float16 if is_half else torch.float32 def get_bert_inf(phones, word2ph, norm_text, language): language = language.replace("all_", "") if language == "zh": bert = get_bert_feature(norm_text, word2ph).to(device) # .to(dtype) else: bert = torch.zeros( (1024, len(phones)), dtype=torch.float16 if is_half else torch.float32, ).to(device) return bert 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 get_phones_and_bert(text, language): if language in {"en", "all_zh", "all_ja"}: language = language.replace("all_", "") if language == "en": LangSegment.setfilters(["en"]) formattext = " ".join(tmp["text"] for tmp in LangSegment.getTexts(text)) else: # 因无法区别中日文汉字,以用户输入为准 formattext = text while " " in formattext: formattext = formattext.replace(" ", " ") phones, word2ph, norm_text = clean_text_inf(formattext, language) if language == "zh": bert = get_bert_feature(norm_text, word2ph).to(device) else: bert = torch.zeros( (1024, len(phones)), dtype=torch.float16 if is_half == True else torch.float32, ).to(device) elif language in {"zh", "ja", "auto"}: textlist = [] langlist = [] LangSegment.setfilters(["zh", "ja", "en", "ko"]) if language == "auto": for tmp in LangSegment.getTexts(text): if tmp["lang"] == "ko": langlist.append("zh") textlist.append(tmp["text"]) else: langlist.append(tmp["lang"]) textlist.append(tmp["text"]) else: for tmp in LangSegment.getTexts(text): if tmp["lang"] == "en": langlist.append(tmp["lang"]) else: # 因无法区别中日文汉字,以用户输入为准 langlist.append(language) textlist.append(tmp["text"]) print(textlist) print(langlist) phones_list = [] bert_list = [] norm_text_list = [] for i in range(len(textlist)): lang = langlist[i] phones, word2ph, norm_text = clean_text_inf(textlist[i], lang) bert = get_bert_inf(phones, word2ph, norm_text, lang) phones_list.append(phones) norm_text_list.append(norm_text) bert_list.append(bert) bert = torch.cat(bert_list, dim=1) phones = sum(phones_list, []) norm_text = ''.join(norm_text_list) return phones, bert.to(dtype), norm_text def set_gpt_weights(gpt_path): global hz, max_sec, t2s_model, config hz = 50 dict_s1 = torch.load(gpt_path, map_location="cpu") config = dict_s1["config"] max_sec = config["data"]["max_sec"] t2s_model = Text2SemanticLightningModule(config, "****", is_train=False) t2s_model.load_state_dict(dict_s1["weight"]) if is_half: 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)) with open("./gweight.txt", "w", encoding="utf-8") as f: f.write(gpt_path) def set_sovits_weights(sovits_path): global vq_model, hps dict_s2 = torch.load(sovits_path, map_location="cpu") hps = dict_s2["config"] hps = DictToAttrRecursive(hps) hps.model.semantic_frame_rate = "25hz" 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 "pretrained" not in sovits_path: del vq_model.enc_q if is_half: vq_model = vq_model.half().to(device) else: vq_model = vq_model.to(device) vq_model.eval() print(vq_model.load_state_dict(dict_s2["weight"], strict=False)) with open("./sweight.txt", "w", encoding="utf-8") as f: f.write(sovits_path) def gen_audio(ref_wav_path, prompt_text, text_to_speak, output_file, top_k=20, top_p=0.6, temperature=0.6, ref_free=False): if prompt_text is None or len(prompt_text) == 0: ref_free = True t0 = ttime() prompt_language = "zh" text_language = "zh" if not ref_free: prompt_text = prompt_text.strip("\n") if prompt_text[-1] not in splits: prompt_text += "。" if prompt_language != "en" else "." print(i18n("实际输入的参考文本:"), prompt_text) text_to_speak = text_to_speak.strip("\n") text_to_speak = replace_consecutive_punctuation(text_to_speak) if text_to_speak[0] not in splits and len(get_first(text_to_speak)) < 4: text_to_speak = "。" + text_to_speak if text_language != "en" else "." + text_to_speak print(i18n("实际输入的目标文本:"), text_to_speak) zero_wav = np.zeros( int(hps.data.sampling_rate * 0.3), dtype=np.float16 if is_half == True else np.float32, ) if not ref_free: with torch.no_grad(): wav16k, sr = librosa.load(ref_wav_path, sr=16000) if wav16k.shape[0] > 160000 or wav16k.shape[0] < 48000: raise OSError(i18n("参考音频在3~10秒范围外,请更换!")) wav16k = torch.from_numpy(wav16k) zero_wav_torch = torch.from_numpy(zero_wav) if is_half: wav16k = wav16k.half().to(device) zero_wav_torch = zero_wav_torch.half().to(device) else: wav16k = wav16k.to(device) zero_wav_torch = zero_wav_torch.to(device) wav16k = torch.cat([wav16k, zero_wav_torch]) ssl_content = ssl_model.model(wav16k.unsqueeze(0))[ "last_hidden_state" ].transpose( 1, 2 ) # .float() codes = vq_model.extract_latent(ssl_content) prompt_semantic = codes[0, 0] prompt = prompt_semantic.unsqueeze(0).to(device) t1 = ttime() # text_to_speak = cut1(text_to_speak) text_to_speak = cut3(text_to_speak) while "\n\n" in text_to_speak: text_to_speak = text_to_speak.replace("\n\n", "\n") print(i18n("实际输入的目标文本(切句后):"), text_to_speak) texts = text_to_speak.split("\n") texts = process_text(texts) texts = merge_short_text_in_array(texts, 5) audio_opt = [] if not ref_free: phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language) for text_to_speak in texts: # 解决输入目标文本的空行导致报错的问题 if len(text_to_speak.strip()) == 0: continue if text_to_speak[-1] not in splits: text_to_speak += "。" if text_language != "en" else "." print(i18n("实际输入的目标文本(每句):"), text_to_speak) phones2, bert2, norm_text2 = get_phones_and_bert(text_to_speak, text_language) print(i18n("前端处理后的文本(每句):"), norm_text2) if not ref_free: bert = torch.cat([bert1, bert2], 1) all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0) else: bert = bert2 all_phoneme_ids = torch.LongTensor(phones2).to(device).unsqueeze(0) bert = bert.to(device).unsqueeze(0) all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device) t2 = ttime() with torch.no_grad(): # pred_semantic = t2s_model.model.infer( pred_semantic, idx = t2s_model.model.infer_panel( all_phoneme_ids, all_phoneme_len, None if ref_free else prompt, bert, # prompt_phone_len=ph_offset, top_k=top_k, top_p=top_p, temperature=temperature, early_stop_num=hz * max_sec, ) t3 = ttime() # print(pred_semantic.shape,idx) pred_semantic = pred_semantic[:, -idx:].unsqueeze( 0 ) # .unsqueeze(0)#mq要多unsqueeze一次 refer = get_spepc(hps, ref_wav_path) # .to(device) if is_half: refer = refer.half().to(device) else: refer = refer.to(device) # audio = vq_model.decode(pred_semantic, all_phoneme_ids, refer).detach().cpu().numpy()[0, 0] audio = ( vq_model.decode( pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refer ) .detach() .cpu() .numpy()[0, 0] ) # 试试重建不带上prompt部分 max_audio = np.abs(audio).max() # 简单防止16bit爆音 if max_audio > 1: audio /= max_audio audio_opt.append(audio) audio_opt.append(zero_wav) t4 = ttime() # 将音频数据合并 audio_data = np.concatenate(audio_opt, 0) * 32768 audio_data = audio_data.astype(np.int16) wavfile.write(output_file, hps.data.sampling_rate, audio_data) print("%.3f\t%.3f\t%.3f\t%.3f" % (t1 - t0, t2 - t1, t3 - t2, t4 - t3)) cnhubert_base_path = os.environ.get( "cnhubert_base_path", "GPT_SoVITS/pretrained_models/chinese-hubert-base" ) bert_path = os.environ.get( "bert_path", "GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large" ) cnhubert.cnhubert_base_path = cnhubert_base_path ssl_model = cnhubert.get_model() if is_half: ssl_model = ssl_model.half().to(device) else: ssl_model = ssl_model.to(device) tokenizer = AutoTokenizer.from_pretrained(bert_path) bert_model = AutoModelForMaskedLM.from_pretrained(bert_path) if is_half: bert_model = bert_model.half().to(device) else: bert_model = bert_model.to(device) def speak(text_to_speak): sovits_path = "SoVITS_weights/阿贝多_e12_s2748.pth" set_sovits_weights(sovits_path) gpt_path = "GPT_weights/阿贝多-e10.ckpt" set_gpt_weights(gpt_path) ref_wav_path = "audio/首先,先看看这不明来源的元素力,究竟是如何对外流动的.wav" prompt_text = "首先,先看看这不明来源的元素力,究竟是如何对外流动的。" # text_to_speak = "我...我...我不知道你在说什么,我们之间没有秘密呀。可能你弄错了,我们平时关系很好的,请不要误会。" # 创建一个时间戳的文件名 output_file = "outputs/" + str(int(ttime())) + ".wav" gen_audio(ref_wav_path, prompt_text, text_to_speak, output_file) return output_file def main(): speak("放学了,我该回家了,你叫我留下来干什么?") if __name__ == '__main__': main()