Mahiruoshi
commited on
Commit
·
c7aa991
1
Parent(s):
2618e11
Update main.py
Browse files
main.py
CHANGED
@@ -19,6 +19,8 @@ import os
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import pickle
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import openai
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from scipy.io.wavfile import write
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def is_japanese(string):
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for ch in string:
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if ord(ch) > 0x3040 and ord(ch) < 0x30FF:
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@@ -42,6 +44,7 @@ def extrac(text):
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i = romajitable.to_kana(i).katakana
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i = i.replace('\n','').replace(' ','')
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#Current length of single sentence: 20
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if len(i)>1:
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if len(i) > 20:
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try:
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@@ -53,6 +56,8 @@ def extrac(text):
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pass
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else:
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final_list.append(i)
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final_list = [x for x in final_list if x != '']
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print(final_list)
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return final_list
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@@ -98,7 +103,7 @@ def get_symbols_from_json(path):
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return data['symbols']
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def sle(language,text):
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text = text.replace('\n', '
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if language == "中文":
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tts_input1 = "[ZH]" + text + "[ZH]"
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return tts_input1
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@@ -124,6 +129,7 @@ def get_text(text,hps_ms):
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def create_tts_fn(net_g,hps,speaker_id):
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speaker_id = int(speaker_id)
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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 ):
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repeat_time = int(repeat_time)
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if is_gpt:
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openai.api_key = api_key
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@@ -182,44 +188,241 @@ def create_tts_fn(net_g,hps,speaker_id):
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print(time_end)
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f1.write(str(c-1)+'\n'+time_start+' --> '+time_end+'\n'+sentence+'\n\n')
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audio_fin.append(audio)
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file_path = "subtitles.srt"
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return history,file_path,(hps.data.sampling_rate, np.concatenate(audio_fin))
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return tts_fn
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def bot(history,user_message):
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return history + [[user_message, None]]
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if __name__ == '__main__':
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hps = utils.get_hparams_from_file('checkpoints/tmp/config.json')
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dev = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
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models = []
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-
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lan = ["中文","日文","自动","手动"]
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with open("checkpoints/info.json", "r", encoding="utf-8") as f:
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models_info = json.load(f)
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checkpoint = models_info['Seisho Music Academy']["checkpoint"]
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phone_dict = {
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symbol: i for i, symbol in enumerate(symbols)
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}
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net_g = SynthesizerTrn(
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len(symbols),
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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).to(dev)
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_ = net_g.eval()
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_ = utils.load_checkpoint(checkpoint, net_g)
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for i in models_info:
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school = models_info[i]
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speakers = school["speakers"]
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content = []
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@@ -230,12 +433,14 @@ if __name__ == '__main__':
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name = speakers[j]["name"]
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content.append((sid, name, title, example, create_tts_fn(net_g,hps,sid)))
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models.append(content)
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-
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with gr.Blocks() as app:
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with gr.Tabs():
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for i in schools:
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with gr.TabItem(i):
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with gr.TabItem(name):
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with gr.Column():
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with gr.Row():
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@@ -255,19 +460,45 @@ if __name__ == '__main__':
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with gr.Accordion(label="Setting", open=False):
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input2 = gr.Dropdown(label="Language", choices=lan, value="自动", interactive=True)
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input3 = gr.Checkbox(value=False, label="长句切割(小说合成)")
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input4 = gr.Slider(minimum=0, maximum=1.0, label="更改噪声比例(noise scale),以控制情感", value=0.
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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.
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input6 = gr.Slider(minimum=0.1, maximum=10, label="duration", value=1)
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with gr.Accordion(label="Advanced Setting", open=False):
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audio_input3 = gr.Dropdown(label="重复次数", choices=list(range(101)), value='0', interactive=True)
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api_input1 = gr.Checkbox(value=False, label="接入chatgpt")
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api_input2 = gr.TextArea(label="api-key",lines=1,value = '
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output2 = gr.outputs.File(label="字幕文件:subtitles.srt")
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audio_input1 = gr.Checkbox(value=False, label="修改音频路径(live2d)")
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audio_input2 = gr.TextArea(label="音频路径",lines=1,value = '
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btnVC.click(bot, inputs = [chatbot,input1], outputs = [chatbot]).then(
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tts_fn, inputs=[chatbot,api_input1,api_input2,audio_input1,audio_input2,audio_input3,input1,input2,input3,input4,input5,input6], outputs=[chatbot,output2,output1]
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)
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-
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app.launch()
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import pickle
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import openai
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from scipy.io.wavfile import write
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import librosa
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from mel_processing import spectrogram_torch
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def is_japanese(string):
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for ch in string:
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if ord(ch) > 0x3040 and ord(ch) < 0x30FF:
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i = romajitable.to_kana(i).katakana
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i = i.replace('\n','').replace(' ','')
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#Current length of single sentence: 20
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'''
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if len(i)>1:
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if len(i) > 20:
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try:
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pass
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else:
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final_list.append(i)
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'''
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final_list.append(i)
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final_list = [x for x in final_list if x != '']
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print(final_list)
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return final_list
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return data['symbols']
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def sle(language,text):
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text = text.replace('\n', '').replace('\r', '').replace(" ", "")
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if language == "中文":
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tts_input1 = "[ZH]" + text + "[ZH]"
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return tts_input1
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def create_tts_fn(net_g,hps,speaker_id):
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speaker_id = int(speaker_id)
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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 ):
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text = check_text(text)
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repeat_time = int(repeat_time)
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if is_gpt:
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openai.api_key = api_key
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print(time_end)
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f1.write(str(c-1)+'\n'+time_start+' --> '+time_end+'\n'+sentence+'\n\n')
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audio_fin.append(audio)
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try:
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write(audiopath + '.wav',22050,np.concatenate(audio_fin))
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if is_audio:
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for i in range(repeat_time):
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cmd = 'ffmpeg -y -i ' + audiopath + '.wav' + ' -ar 44100 '+ audiopath.replace('temp','temp'+str(i))
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os.system(cmd)
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except:
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pass
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file_path = "subtitles.srt"
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return history,file_path,(hps.data.sampling_rate, np.concatenate(audio_fin))
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return tts_fn
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def create_vc_fn(net_g,hps):
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def vc_fn(text,language,n_scale,n_scale_w,l_scale,original_speaker, target_speaker, record_audio, upload_audio):
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input_audio = record_audio if record_audio is not None else upload_audio
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original_speaker_id = selection(original_speaker)
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target_speaker_id = selection(target_speaker)
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if input_audio is None:
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stn_tst = get_text(sle(language,text),hps)
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with torch.no_grad():
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x_tst = stn_tst.unsqueeze(0).to(dev)
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x_tst_lengths = torch.LongTensor([stn_tst.size(0)]).to(dev)
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sid = torch.LongTensor([original_speaker_id]).to(dev)
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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()
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sampling_rate = hps.data.sampling_rate
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else:
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sampling_rate, audio = input_audio
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audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
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if len(audio.shape) > 1:
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audio = librosa.to_mono(audio.transpose(1, 0))
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if sampling_rate != hps.data.sampling_rate:
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audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=hps.data.sampling_rate)
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with torch.no_grad():
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y = torch.FloatTensor(audio)
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y = y / max(-y.min(), y.max()) / 0.99
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y = y.to(dev)
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y = y.unsqueeze(0)
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spec = spectrogram_torch(y, hps.data.filter_length,
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hps.data.sampling_rate, hps.data.hop_length, hps.data.win_length,
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center=False).to(dev)
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spec_lengths = torch.LongTensor([spec.size(-1)]).to(dev)
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sid_src = torch.LongTensor([original_speaker_id]).to(dev)
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sid_tgt = torch.LongTensor([target_speaker_id]).to(dev)
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audio = net_g.voice_conversion(spec, spec_lengths, sid_src=sid_src, sid_tgt=sid_tgt)[0][
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0, 0].data.cpu().float().numpy()
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del y, spec, spec_lengths, sid_src, sid_tgt
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return "Success", (hps.data.sampling_rate, audio)
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return vc_fn
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def bot(history,user_message):
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return history + [[check_text(user_message), None]]
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def selection(speaker):
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if speaker == "高咲侑":
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spk = 0
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return spk
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elif speaker == "歩夢":
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spk = 1
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return spk
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elif speaker == "かすみ":
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spk = 2
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return spk
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elif speaker == "しずく":
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spk = 3
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return spk
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elif speaker == "果林":
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spk = 4
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return spk
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elif speaker == "愛":
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spk = 5
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return spk
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elif speaker == "彼方":
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spk = 6
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return spk
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elif speaker == "せつ菜":
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spk = 7
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return spk
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elif speaker == "エマ":
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spk = 8
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return spk
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elif speaker == "璃奈":
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spk = 9
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return spk
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elif speaker == "栞子":
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spk = 10
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return spk
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elif speaker == "ランジュ":
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spk = 11
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return spk
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elif speaker == "ミア":
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spk = 12
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return spk
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elif speaker == "派蒙":
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spk = 16
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return spk
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elif speaker == "c1":
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spk = 18
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return spk
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elif speaker == "c2":
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spk = 19
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return spk
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elif speaker == "華恋":
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spk = 21
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return spk
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elif speaker == "まひる":
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spk = 22
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return spk
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elif speaker == "なな":
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spk = 23
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return spk
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elif speaker == "クロディーヌ":
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spk = 24
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return spk
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elif speaker == "ひかり":
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spk = 25
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return spk
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elif speaker == "純那":
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spk = 26
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return spk
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elif speaker == "香子":
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spk = 27
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return spk
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elif speaker == "真矢":
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spk = 28
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return spk
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elif speaker == "双葉":
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spk = 29
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return spk
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elif speaker == "ミチル":
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spk = 30
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return spk
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elif speaker == "メイファン":
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spk = 31
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return spk
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elif speaker == "やちよ":
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spk = 32
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return spk
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elif speaker == "晶":
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spk = 33
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return spk
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elif speaker == "いちえ":
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spk = 34
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return spk
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elif speaker == "ゆゆ子":
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spk = 35
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return spk
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elif speaker == "塁":
|
371 |
+
spk = 36
|
372 |
+
return spk
|
373 |
+
|
374 |
+
elif speaker == "珠緒":
|
375 |
+
spk = 37
|
376 |
+
return spk
|
377 |
+
|
378 |
+
elif speaker == "あるる":
|
379 |
+
spk = 38
|
380 |
+
return spk
|
381 |
+
|
382 |
+
elif speaker == "ララフィン":
|
383 |
+
spk = 39
|
384 |
+
return spk
|
385 |
+
|
386 |
+
elif speaker == "美空":
|
387 |
+
spk = 40
|
388 |
+
return spk
|
389 |
+
|
390 |
+
elif speaker == "静羽":
|
391 |
+
spk = 41
|
392 |
+
return spk
|
393 |
+
|
394 |
+
else:
|
395 |
+
return 0
|
396 |
+
|
397 |
+
def check_text(input):
|
398 |
+
if isinstance(input, str):
|
399 |
+
return input
|
400 |
+
else:
|
401 |
+
with open(input.name, "r", encoding="utf-8") as f:
|
402 |
+
return f.read()
|
403 |
|
404 |
if __name__ == '__main__':
|
405 |
hps = utils.get_hparams_from_file('checkpoints/tmp/config.json')
|
406 |
dev = torch.device("cuda:0" if torch.cuda.is_available() else "cpu")
|
407 |
models = []
|
408 |
+
schools_list = ["ShojoKageki-Nijigasaki","ShojoKageki","Nijigasaki"]
|
409 |
+
schools = []
|
410 |
lan = ["中文","日文","自动","手动"]
|
411 |
with open("checkpoints/info.json", "r", encoding="utf-8") as f:
|
412 |
models_info = json.load(f)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
413 |
for i in models_info:
|
414 |
+
checkpoint = models_info[i]["checkpoint"]
|
415 |
+
phone_dict = {
|
416 |
+
symbol: i for i, symbol in enumerate(symbols)
|
417 |
+
}
|
418 |
+
net_g = SynthesizerTrn(
|
419 |
+
len(symbols),
|
420 |
+
hps.data.filter_length // 2 + 1,
|
421 |
+
hps.train.segment_size // hps.data.hop_length,
|
422 |
+
n_speakers=hps.data.n_speakers,
|
423 |
+
**hps.model).to(dev)
|
424 |
+
_ = net_g.eval()
|
425 |
+
_ = utils.load_checkpoint(checkpoint, net_g)
|
426 |
school = models_info[i]
|
427 |
speakers = school["speakers"]
|
428 |
content = []
|
|
|
433 |
name = speakers[j]["name"]
|
434 |
content.append((sid, name, title, example, create_tts_fn(net_g,hps,sid)))
|
435 |
models.append(content)
|
436 |
+
schools.append((i,create_vc_fn(net_g,hps)))
|
437 |
with gr.Blocks() as app:
|
438 |
with gr.Tabs():
|
439 |
+
for (i,vc_fn) in schools:
|
440 |
with gr.TabItem(i):
|
441 |
+
idols = ["派蒙"]
|
442 |
+
for (sid, name, title, example, tts_fn) in models[schools_list.index(i)]:
|
443 |
+
idols.append(name)
|
444 |
with gr.TabItem(name):
|
445 |
with gr.Column():
|
446 |
with gr.Row():
|
|
|
460 |
with gr.Accordion(label="Setting", open=False):
|
461 |
input2 = gr.Dropdown(label="Language", choices=lan, value="自动", interactive=True)
|
462 |
input3 = gr.Checkbox(value=False, label="长句切割(小说合成)")
|
463 |
+
input4 = gr.Slider(minimum=0, maximum=1.0, label="更改噪声比例(noise scale),以控制情感", value=0.6)
|
464 |
+
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=0.668)
|
465 |
input6 = gr.Slider(minimum=0.1, maximum=10, label="duration", value=1)
|
466 |
with gr.Accordion(label="Advanced Setting", open=False):
|
467 |
audio_input3 = gr.Dropdown(label="重复次数", choices=list(range(101)), value='0', interactive=True)
|
468 |
api_input1 = gr.Checkbox(value=False, label="接入chatgpt")
|
469 |
+
api_input2 = gr.TextArea(label="api-key",lines=1,value = '懂得都懂')
|
470 |
output2 = gr.outputs.File(label="字幕文件:subtitles.srt")
|
471 |
audio_input1 = gr.Checkbox(value=False, label="修改音频路径(live2d)")
|
472 |
+
audio_input2 = gr.TextArea(label="音频路径",lines=1,value = 'D:/path/to/live2d/sounds/temp.wav')
|
473 |
+
input3 = gr.Checkbox(value=False, label="长句切割(小说合成)")
|
474 |
+
inputxt = gr.File(label="Text")
|
475 |
+
btnbook = gr.Button("小说合成")
|
476 |
btnVC.click(bot, inputs = [chatbot,input1], outputs = [chatbot]).then(
|
477 |
tts_fn, inputs=[chatbot,api_input1,api_input2,audio_input1,audio_input2,audio_input3,input1,input2,input3,input4,input5,input6], outputs=[chatbot,output2,output1]
|
478 |
)
|
479 |
+
btnbook.click(bot, inputs = [chatbot,inputxt], outputs = [chatbot]).then(
|
480 |
+
tts_fn, inputs=[chatbot,api_input1,api_input2,audio_input1,audio_input2,audio_input3,inputxt,input2,input3,input4,input5,input6], outputs=[chatbot,output2,output1]
|
481 |
+
)
|
482 |
+
with gr.Tab("Voice Conversion(类似sovits)"):
|
483 |
+
gr.Markdown("""
|
484 |
+
声线转化,使用模型中的说话人作为音源时效果更佳
|
485 |
+
""")
|
486 |
+
with gr.Column():
|
487 |
+
with gr.Accordion(label="方法1:录制或上传声音,可进行歌声合成", open=False):
|
488 |
+
record_audio = gr.Audio(label="record your voice", source="microphone")
|
489 |
+
upload_audio = gr.Audio(label="or upload audio here", source="upload")
|
490 |
+
with gr.Accordion(label="方法2:由原说话人先进行tts后套娃,适用于合成中文等特殊场景", open=True):
|
491 |
+
text = gr.TextArea(label="Text", value='由源说话人进行语音转化',lines = 1)
|
492 |
+
language = gr.Dropdown(label="Language", choices=lan, value="自动", interactive=True)
|
493 |
+
n_scale = gr.Slider(minimum=0, maximum=1.0, label="更改噪声比例(noise scale),以控制情感", value=0.6)
|
494 |
+
n_scale_w = 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=0.668)
|
495 |
+
l_scale = gr.Slider(minimum=0.1, maximum=10, label="duration", value=1.1)
|
496 |
+
source_speaker = gr.Dropdown(choices=idols, value=idols[-2], label="source speaker")
|
497 |
+
target_speaker = gr.Dropdown(choices=idols, value=idols[-3], label="target speaker")
|
498 |
+
with gr.Column():
|
499 |
+
message_box = gr.Textbox(label="Message")
|
500 |
+
converted_audio = gr.Audio(label='converted audio')
|
501 |
+
btn = gr.Button("Convert!")
|
502 |
+
btn.click(vc_fn, inputs=[text,language,n_scale,n_scale_w,l_scale,source_speaker, target_speaker, record_audio, upload_audio],
|
503 |
+
outputs=[message_box, converted_audio])
|
504 |
app.launch()
|