import logging logging.getLogger('numba').setLevel(logging.WARNING) logging.getLogger('matplotlib').setLevel(logging.WARNING) logging.getLogger('urllib3').setLevel(logging.WARNING) import json import re import numpy as np import IPython.display as ipd import torch import commons import utils from models import SynthesizerTrn from text.symbols import symbols from text import text_to_sequence import gradio as gr import time import datetime import os import pickle import openai from scipy.io.wavfile import write def is_japanese(string): for ch in string: if ord(ch) > 0x3040 and ord(ch) < 0x30FF: return True return False def is_english(string): import re pattern = re.compile('^[A-Za-z0-9.,:;!?()_*"\' ]+$') if pattern.fullmatch(string): return True else: return False def extrac(text): text = re.sub("<[^>]*>","",text) result_list = re.split(r'\n', text) final_list = [] for i in result_list: if is_english(i): i = romajitable.to_kana(i).katakana i = i.replace('\n','').replace(' ','') #Current length of single sentence: 20 if len(i)>1: if len(i) > 20: try: cur_list = re.split(r'。|!', i) for i in cur_list: if len(i)>1: final_list.append(i+'。') except: pass else: final_list.append(i) final_list = [x for x in final_list if x != ''] print(final_list) return final_list def to_numpy(tensor: torch.Tensor): return tensor.detach().cpu().numpy() if tensor.requires_grad \ else tensor.detach().numpy() def chatgpt(text): messages = [] try: if text != 'exist': with open('log.pickle', 'rb') as f: messages = pickle.load(f) messages.append({"role": "user", "content": text},) chat = openai.ChatCompletion.create(model="gpt-3.5-turbo", messages=messages) reply = chat.choices[0].message.content messages.append({"role": "assistant", "content": reply}) print(messages[-1]) if len(messages) == 12: messages[6:10] = messages[8:] del messages[-2:] with open('log.pickle', 'wb') as f: pickle.dump(messages, f) return reply except: messages.append({"role": "user", "content": text},) chat = openai.ChatCompletion.create(model="gpt-3.5-turbo", messages=messages) reply = chat.choices[0].message.content messages.append({"role": "assistant", "content": reply}) print(messages[-1]) if len(messages) == 12: messages[6:10] = messages[8:] del messages[-2:] with open('log.pickle', 'wb') as f: pickle.dump(messages, f) return reply def get_symbols_from_json(path): assert os.path.isfile(path) with open(path, 'r') as f: data = json.load(f) return data['symbols'] def sle(language,text): text = text.replace('\n', ' ').replace('\r', '').replace(" ", "") if language == "中文": tts_input1 = "[ZH]" + text + "[ZH]" return tts_input1 elif language == "自动": tts_input1 = f"[JA]{text}[JA]" if is_japanese(text) else f"[ZH]{text}[ZH]" return tts_input1 elif language == "日文": tts_input1 = "[JA]" + text + "[JA]" return tts_input1 elif language == "英文": tts_input1 = "[EN]" + text + "[EN]" return tts_input1 elif language == "手动": return text def get_text(text,hps_ms): text_norm = text_to_sequence(text,hps_ms.data.text_cleaners) if hps_ms.data.add_blank: text_norm = commons.intersperse(text_norm, 0) text_norm = torch.LongTensor(text_norm) return text_norm def create_tts_fn(net_g,hps,speaker_id): speaker_id = int(speaker_id) def tts_fn(history,is_gpt,api_key,is_audio,audiopath,repeat_time,text, language, extract, n_scale= 0.667,n_scale_w = 0.8, l_scale = 1 ): repeat_time = int(repeat_time) if is_gpt: openai.api_key = api_key text = chatgpt(text) history[-1][1] = text if not extract: print(text) t1 = time.time() stn_tst = get_text(sle(language,text),hps) with torch.no_grad(): x_tst = stn_tst.unsqueeze(0).to(dev) x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(dev) sid = torch.LongTensor([speaker_id]).to(dev) audio = net_g.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=n_scale, noise_scale_w=n_scale_w, length_scale=l_scale)[0][0,0].data.cpu().float().numpy() t2 = time.time() spending_time = "推理时间为:"+str(t2-t1)+"s" print(spending_time) file_path = "subtitles.srt" try: write(audiopath + '.wav',22050,audio) if is_audio: for i in range(repeat_time): cmd = 'ffmpeg -y -i ' + audiopath + '.wav' + ' -ar 44100 '+ audiopath.replace('temp','temp'+str(i)) os.system(cmd) except: pass return history,file_path,(hps.data.sampling_rate,audio) else: a = ['【','[','(','('] b = ['】',']',')',')'] for i in a: text = text.replace(i,'<') for i in b: text = text.replace(i,'>') final_list = extrac(text.replace('“','').replace('”','')) audio_fin = [] c = 0 t = datetime.timedelta(seconds=0) f1 = open("subtitles.srt",'w',encoding='utf-8') for sentence in final_list: c +=1 stn_tst = get_text(sle(language,sentence),hps) with torch.no_grad(): x_tst = stn_tst.unsqueeze(0).to(dev) x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(dev) sid = torch.LongTensor([speaker_id]).to(dev) t1 = time.time() audio = net_g.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=n_scale, noise_scale_w=n_scale_w, length_scale=l_scale)[0][0,0].data.cpu().float().numpy() t2 = time.time() spending_time = "第"+str(c)+"句的推理时间为:"+str(t2-t1)+"s" print(spending_time) time_start = str(t).split(".")[0] + "," + str(t.microseconds)[:3] last_time = datetime.timedelta(seconds=len(audio)/float(22050)) t+=last_time time_end = str(t).split(".")[0] + "," + str(t.microseconds)[:3] print(time_end) f1.write(str(c-1)+'\n'+time_start+' --> '+time_end+'\n'+sentence+'\n\n') audio_fin.append(audio) try: write(audiopath + '.wav',22050,np.concatenate(audio_fin)) if is_audio: for i in range(repeat_time): cmd = 'ffmpeg -y -i ' + audiopath + '.wav' + ' -ar 44100 '+ audiopath.replace('temp','temp'+str(i)) os.system(cmd) except: pass file_path = "subtitles.srt" return history,file_path,(hps.data.sampling_rate, np.concatenate(audio_fin)) return tts_fn def bot(history,user_message): return history + [[user_message, None]] if __name__ == '__main__': hps = utils.get_hparams_from_file('checkpoints/tmp/config.json') dev = torch.device("cuda:0" if torch.cuda.is_available() else "cpu") models = [] schools = ["Nijigasaki High School","Seisho Music Academy","Rinmeikan Girls School","Frontier School of Arts","Siegfeld Institute of Music"] lan = ["中文","日文","自动","手动"] with open("checkpoints/info.json", "r", encoding="utf-8") as f: models_info = json.load(f) checkpoint = models_info['Seisho Music Academy']["checkpoint"] phone_dict = { symbol: i for i, symbol in enumerate(symbols) } net_g = SynthesizerTrn( len(symbols), hps.data.filter_length // 2 + 1, hps.train.segment_size // hps.data.hop_length, n_speakers=hps.data.n_speakers, **hps.model).to(dev) _ = net_g.eval() _ = utils.load_checkpoint(checkpoint, net_g) for i in models_info: school = models_info[i] speakers = school["speakers"] content = [] for j in speakers: sid = int(speakers[j]['sid']) title = school example = speakers[j]['speech'] name = speakers[j]["name"] content.append((sid, name, title, example, create_tts_fn(net_g,hps,sid))) models.append(content) with gr.Blocks() as app: with gr.Tabs(): for i in schools: with gr.TabItem(i): for (sid, name, title, example, tts_fn) in models[schools.index(i)]: with gr.TabItem(name): with gr.Column(): with gr.Row(): with gr.Row(): gr.Markdown( '
' f'' '
' ) chatbot = gr.Chatbot() with gr.Row(): with gr.Column(scale=0.85): input1 = gr.TextArea(label="Text", value=example,lines = 1) with gr.Column(scale=0.15, min_width=0): btnVC = gr.Button("Send") output1 = gr.Audio(label="采样率22050") with gr.Accordion(label="Setting", open=False): input2 = gr.Dropdown(label="Language", choices=lan, value="自动", interactive=True) input3 = gr.Checkbox(value=False, label="长句切割(小说合成)") input4 = gr.Slider(minimum=0, maximum=1.0, label="更改噪声比例(noise scale),以控制情感", value=0.567) input5 = gr.Slider(minimum=0, maximum=1.0, label="更改噪声偏差(noise scale w),以控制音素长短", value=0.7) input6 = gr.Slider(minimum=0.1, maximum=10, label="duration", value=1) with gr.Accordion(label="Advanced Setting", open=False): audio_input3 = gr.Dropdown(label="重复次数", choices=list(range(101)), value='0', interactive=True) api_input1 = gr.Checkbox(value=False, label="接入chatgpt") api_input2 = gr.TextArea(label="api-key",lines=1,value = '见 https://openai.com/blog/openai-api') output2 = gr.outputs.File(label="字幕文件:subtitles.srt") audio_input1 = gr.Checkbox(value=False, label="修改音频路径(live2d)") audio_input2 = gr.TextArea(label="音频路径",lines=1,value = '#参考 D:/app_develop/live2d_whole/2010002/sounds/temp.wav') btnVC.click(bot, inputs = [chatbot,input1], outputs = [chatbot]).then( tts_fn, inputs=[chatbot,api_input1,api_input2,audio_input1,audio_input2,audio_input3,input1,input2,input3,input4,input5,input6], outputs=[chatbot,output2,output1] ) app.launch()