Lovelive_Nijigasaki_VITS / inference_ork.py
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Update inference_ork.py
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#basic enviornments & openai
import romajitable
import re
import os
import numpy as np
import logging
logging.getLogger('numba').setLevel(logging.WARNING)
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 openai
import tkinter as tk
from tkinter import scrolledtext
import argparse
import time
from scipy.io.wavfile import write
def get_args():
parser = argparse.ArgumentParser(description='inference')
parser.add_argument('--model', default = 'lovelive/G_936000.pth')
parser.add_argument('--audio',
type=str,
help='the sound file of live2d to be replace,assuming they are temp1.wav,temp2.wav,temp3.wav......',
default = 'path/to/temp.wav')
parser.add_argument('--cfg', default="lovelive/config.json")
parser.add_argument('--key',default = "openai key",
help='platform.openai.com')
args = parser.parse_args()
return args
args = get_args()
dev = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
dev = torch.device("cuda:0")
hps_ms = utils.get_hparams_from_file(args.cfg)
#mult-speakers
net_g_ms = SynthesizerTrn(
len(symbols),
hps_ms.data.filter_length // 2 + 1,
hps_ms.train.segment_size // hps_ms.data.hop_length,
n_speakers=hps_ms.data.n_speakers,
**hps_ms.model).to(dev)
_ = net_g_ms.eval()
_ = utils.load_checkpoint(args.model, net_g_ms, None)
# detecting japanese
def is_japanese(string):
for ch in string:
if ord(ch) > 0x3040 and ord(ch) < 0x30FF:
return True
return False
def get_text(text, hps):
text_norm = text_to_sequence(text, hps.data.text_cleaners)
if hps.data.add_blank:
text_norm = commons.intersperse(text_norm, 0)
text_norm = torch.LongTensor(text_norm)
return text_norm
def ttv(text):
text = text.replace('\n','').replace(' ','')
text = f"[JA]{text}[JA]" if is_japanese(text) else f"[ZH]{text}[ZH]"
speaker_id = 7
stn_tst = get_text(text,hps_ms)
t1 = time.time()
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_ms.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=0.467, noise_scale_w=0.5, length_scale=1)[0][0,0].data.cpu().float().numpy()
write(args.audio + '.wav',22050,audio)
i = 0
while i < 19:
i +=1
cmd = 'ffmpeg -y -i ' + args.audio + '.wav' + ' -ar 44100 '+ args.audio.replace('temp','temp'+str(i))
os.system(cmd)
t2 = time.time()
print("推理耗时:",(t2 - t1),"s")
openai.api_key = args.key
result_list = []
messages = []
read_log = input('Loading log?(y/n)')
if read_log == 'y':
messages = []
with open('log.pickle', 'rb') as f:
messages = pickle.load(f)
print('Most recently log:\n'+str(messages[-1]))
def send_message():
text = input_box.get("1.0", "end-1c") # 获取用户输入的文本
messages.append({"role": "user", "content": text},)
chat = openai.ChatCompletion.create(model="gpt-3.5-turbo", messages=messages)
reply = chat.choices[0].message.content
ttv(reply)
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)
chat_box.configure(state='normal')
chat_box.insert(tk.END, "You: " + text + "\n")
chat_box.insert(tk.END, "Tamao: " + reply + "\n")
chat_box.configure(state='disabled')
input_box.delete("1.0", tk.END)
root = tk.Tk()
root.title("Tamao")
chat_box = scrolledtext.ScrolledText(root, width=50, height=10)
chat_box.configure(state='disabled')
chat_box.pack(side=tk.TOP, fill=tk.BOTH, padx=10, pady=10, expand=True)
input_frame = tk.Frame(root)
input_frame.pack(side=tk.BOTTOM, fill=tk.X, padx=10, pady=10)
input_box = tk.Text(input_frame, height=3, width=50)
input_box.pack(side=tk.LEFT, fill=tk.X, padx=10, expand=True)
send_button = tk.Button(input_frame, text="Send", command=send_message)
send_button.pack(side=tk.RIGHT, padx=10)
root.mainloop()