File size: 10,940 Bytes
c56c253
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
4a72c35
 
 
 
 
 
 
 
 
c56c253
 
 
 
 
 
 
 
 
 
 
4adf448
 
 
 
 
 
 
 
 
c56c253
 
 
 
4a72c35
c56c253
 
 
 
 
4adf448
c56c253
 
 
 
 
 
 
 
 
4adf448
 
 
 
 
 
 
 
 
 
 
c56c253
 
 
 
 
 
4a72c35
c56c253
4a72c35
 
c56c253
4a72c35
4adf448
4a72c35
4adf448
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
61af3b3
 
 
 
 
 
 
 
 
 
1b7bfce
4adf448
1b7bfce
 
 
61af3b3
1b7bfce
4adf448
1b7bfce
 
 
4adf448
1b7bfce
 
4adf448
1b7bfce
 
 
4adf448
 
 
 
7fde70a
4adf448
52d6024
 
 
4adf448
 
 
 
 
 
7fde70a
4adf448
 
 
 
 
 
7fde70a
4adf448
 
 
 
 
 
c56c253
4adf448
13f256f
4adf448
 
 
 
 
7fde70a
4adf448
7fde70a
 
4adf448
 
 
c56c253
7fde70a
4adf448
 
 
 
 
 
 
 
c56c253
b9545e1
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
import os
import torch
import librosa
import gradio as gr
from scipy.io.wavfile import write
from transformers import WavLMModel

import utils
from models import SynthesizerTrn
from mel_processing import mel_spectrogram_torch
from speaker_encoder.voice_encoder import SpeakerEncoder

'''
def get_wavlm():
    os.system('gdown https://drive.google.com/uc?id=12-cB34qCTvByWT-QtOcZaqwwO21FLSqU')
    shutil.move('WavLM-Large.pt', 'wavlm')
'''

device = torch.device("cuda" if torch.cuda.is_available() else "cpu")

print("Loading FreeVC...")
hps = utils.get_hparams_from_file("configs/freevc.json")
freevc = SynthesizerTrn(
    hps.data.filter_length // 2 + 1,
    hps.train.segment_size // hps.data.hop_length,
    **hps.model).to(device)
_ = freevc.eval()
_ = utils.load_checkpoint("checkpoints/freevc.pth", freevc, None)
smodel = SpeakerEncoder('speaker_encoder/ckpt/pretrained_bak_5805000.pt')

print("Loading FreeVC(24k)...")
hps = utils.get_hparams_from_file("configs/freevc-24.json")
freevc_24 = SynthesizerTrn(
    hps.data.filter_length // 2 + 1,
    hps.train.segment_size // hps.data.hop_length,
    **hps.model).to(device)
_ = freevc_24.eval()
_ = utils.load_checkpoint("checkpoints/freevc-24.pth", freevc_24, None)

print("Loading FreeVC-s...")
hps = utils.get_hparams_from_file("configs/freevc-s.json")
freevc_s = SynthesizerTrn(
    hps.data.filter_length // 2 + 1,
    hps.train.segment_size // hps.data.hop_length,
    **hps.model).to(device)
_ = freevc_s.eval()
_ = utils.load_checkpoint("checkpoints/freevc-s.pth", freevc_s, None)

print("Loading WavLM for content...")
cmodel = WavLMModel.from_pretrained("microsoft/wavlm-large").to(device)


from openai import OpenAI

import ffmpeg


def convert(api_key, text, tgt, voice, save_path):
    model = "FreeVC (24kHz)"
    with torch.no_grad():
        # tgt
        wav_tgt, _ = librosa.load(tgt, sr=hps.data.sampling_rate)
        wav_tgt, _ = librosa.effects.trim(wav_tgt, top_db=20)
        if model == "FreeVC" or model == "FreeVC (24kHz)":
            g_tgt = smodel.embed_utterance(wav_tgt)
            g_tgt = torch.from_numpy(g_tgt).unsqueeze(0).to(device)
        else:
            wav_tgt = torch.from_numpy(wav_tgt).unsqueeze(0).to(device)
            mel_tgt = mel_spectrogram_torch(
                wav_tgt,
                hps.data.filter_length,
                hps.data.n_mel_channels,
                hps.data.sampling_rate,
                hps.data.hop_length,
                hps.data.win_length,
                hps.data.mel_fmin,
                hps.data.mel_fmax
            )
        # src
        client = OpenAI(api_key=api_key)

        response = client.audio.speech.create(
            model="tts-1-hd",
            voice=voice,
            input=text,
        )

        response.stream_to_file("output_openai.mp3")

        src = "output_openai.mp3"
        wav_src, _ = librosa.load(src, sr=hps.data.sampling_rate)
        wav_src = torch.from_numpy(wav_src).unsqueeze(0).to(device)
        c = cmodel(wav_src).last_hidden_state.transpose(1, 2).to(device)
        # infer
        if model == "FreeVC":
            audio = freevc.infer(c, g=g_tgt)
        elif model == "FreeVC-s":
            audio = freevc_s.infer(c, mel=mel_tgt)
        else:
            audio = freevc_24.infer(c, g=g_tgt)
        audio = audio[0][0].data.cpu().float().numpy()
        if model == "FreeVC" or model == "FreeVC-s":
            write(f"output/{save_path}.wav", hps.data.sampling_rate, audio)
        else:
            write(f"output/{save_path}.wav", 24000, audio)
    return f"output/{save_path}.wav"


class subtitle:
    def __init__(self,index:int, start_time, end_time, text:str):
        self.index = int(index)
        self.start_time = start_time
        self.end_time = end_time
        self.text = text.strip()
    def normalize(self,ntype:str,fps=30):
         if ntype=="prcsv":
              h,m,s,fs=(self.start_time.replace(';',':')).split(":")#seconds
              self.start_time=int(h)*3600+int(m)*60+int(s)+round(int(fs)/fps,2)
              h,m,s,fs=(self.end_time.replace(';',':')).split(":")
              self.end_time=int(h)*3600+int(m)*60+int(s)+round(int(fs)/fps,2)
         elif ntype=="srt":
             h,m,s=self.start_time.split(":")
             s=s.replace(",",".")
             self.start_time=int(h)*3600+int(m)*60+round(float(s),2)
             h,m,s=self.end_time.split(":")
             s=s.replace(",",".")
             self.end_time=int(h)*3600+int(m)*60+round(float(s),2)
         else:
             raise ValueError
    def add_offset(self,offset=0):
        self.start_time+=offset
        if self.start_time<0:
            self.start_time=0
        self.end_time+=offset
        if self.end_time<0:
            self.end_time=0
    def __str__(self) -> str:
        return f'id:{self.index},start:{self.start_time},end:{self.end_time},text:{self.text}'

def read_srt(uploaded_file):
    offset=0
    with open(uploaded_file.name,"r",encoding="utf-8") as f:
        file=f.readlines()
    subtitle_list=[]
    indexlist=[]
    filelength=len(file)
    for i in range(0,filelength):
        if " --> " in file[i]:
            is_st=True
            for char in file[i-1].strip().replace("\ufeff",""):
                if char not in ['0','1','2','3','4','5','6','7','8','9']:
                    is_st=False
                    break
            if is_st:
                indexlist.append(i) #get line id
    listlength=len(indexlist)
    for i in range(0,listlength-1):
        st,et=file[indexlist[i]].split(" --> ")
        id=int(file[indexlist[i]-1].strip().replace("\ufeff",""))
        text=""
        for x in range(indexlist[i]+1,indexlist[i+1]-2):
            text+=file[x]
        st=subtitle(id,st,et,text)
        st.normalize(ntype="srt")
        st.add_offset(offset=offset)
        subtitle_list.append(st)
    st,et=file[indexlist[-1]].split(" --> ")
    id=file[indexlist[-1]-1]
    text=""
    for x in range(indexlist[-1]+1,filelength):
        text+=file[x]
    st=subtitle(id,st,et,text)
    st.normalize(ntype="srt")
    st.add_offset(offset=offset)
    subtitle_list.append(st)
    return subtitle_list

from pydub import AudioSegment

def trim_audio(intervals, input_file_path, output_file_path):
    # load the audio file
    audio = AudioSegment.from_file(input_file_path)

    # iterate over the list of time intervals
    for i, (start_time, end_time) in enumerate(intervals):
        # extract the segment of the audio
        segment = audio[start_time*1000:end_time*1000]

        # construct the output file path
        output_file_path_i = f"{output_file_path}_{i}.wav"

        # export the segment to a file
        segment.export(output_file_path_i, format='wav')

import re

def sort_key(file_name):
    """Extract the last number in the file name for sorting."""
    numbers = re.findall(r'\d+', file_name)
    if numbers:
        return int(numbers[-1])
    return -1  # In case there's no number, this ensures it goes to the start.


def merge_audios(folder_path):
    output_file = "AI配音版.wav"
    # Get all WAV files in the folder
    files = [f for f in os.listdir(folder_path) if f.endswith('.wav')]
    # Sort files based on the last digit in their names
    sorted_files = sorted(files, key=sort_key)
    
    # Initialize an empty audio segment
    merged_audio = AudioSegment.empty()
    
    # Loop through each file, in order, and concatenate them
    for file in sorted_files:
        audio = AudioSegment.from_wav(os.path.join(folder_path, file))
        merged_audio += audio
        print(f"Merged: {file}")
    
    # Export the merged audio to a new file
    merged_audio.export(output_file, format="wav")
    return "AI配音版.wav"

import shutil

def convert_from_srt(apikey, filename, audio_full, voice, multilingual):
    subtitle_list = read_srt(filename)
    audio_data, sr = librosa.load(audio_full, sr=16000, mono=True)

    write("audio_full.wav", sr, audio_data.astype(np.int16))

    if os.path.isdir("output"):
        shutil.rmtree("output")
    if multilingual==False:
        for i in subtitle_list:
            os.makedirs("output", exist_ok=True)
            trim_audio([[i.start_time, i.end_time]], "audio_full.wav", f"sliced_audio_{i.index}")
            print(f"正在合成第{i.index}条语音")
            print(f"语音内容:{i.text}")
            convert(apikey, i.text, f"sliced_audio_{i.index}_0.wav", voice, i.text + " " + str(i.index))
    else:
        for i in subtitle_list:
            os.makedirs("output", exist_ok=True)
            trim_audio([[i.start_time, i.end_time]], "audio_full.wav", f"sliced_audio_{i.index}")
            print(f"正在合成第{i.index}条语音")
            print(f"语音内容:{i.text.splitlines()[1]}")
            convert(apikey, i.text.splitlines()[1], f"sliced_audio_{i.index}_0.wav", voice, i.text.splitlines()[1] + " " + str(i.index))
     
    return merge_audios("output")


with gr.Blocks() as app:
    gr.Markdown("# <center>🌊💕🎶 OpenAI TTS - SRT文件一键AI配音</center>")
    gr.Markdown("### <center>🌟 只需上传SRT文件和原版配音文件即可,每次一集视频AI自动配音!Developed by Kevin Wang </center>")
    with gr.Row():
        with gr.Column():
            inp0 = gr.Textbox(type='password', label='请输入您的OpenAI API Key')
            inp1 = gr.File(file_count="single", label="请上传一集视频对应的SRT文件")
            inp2 = gr.Audio(label="请上传一集视频的配音文件", info="需要是.wav音频文件")
            inp3 = gr.Dropdown(choices=['alloy', 'echo', 'fable', 'onyx', 'nova', 'shimmer'], label='请选择一个说话人提供基础音色', info="试听音色链接:https://platform.openai.com/docs/guides/text-to-speech/voice-options", value='alloy')
            #inp4 = gr.Dropdown(label="请选择用于分离伴奏的模型", info="UVR-HP5去除背景音乐效果更好,但会对人声造成一定的损伤", choices=["UVR-HP2", "UVR-HP5"], value="UVR-HP5")
            inp4 = gr.Checkbox(label="SRT文件是否为双语字幕", info="若为双语字幕,请打勾选择(SRT文件中需要先出现中文字幕,后英文字幕;中英字幕各占一行)")
            btn = gr.Button("一键开启AI配音吧💕", variant="primary")
        with gr.Column():
            out1 = gr.Audio(label="为您生成的AI完整配音", type="filepath")

        btn.click(convert_from_srt, [inp0, inp1, inp2, inp3, inp4], [out1])
        
    gr.Markdown("### <center>注意❗:请勿生成会对任何个人或组织造成侵害的内容,请尊重他人的著作权和知识产权。用户对此程序的任何使用行为与程序开发者无关。</center>")
    gr.HTML('''
        <div class="footer">
                    <p>🌊🏞️🎶 - 江水东流急,滔滔无尽声。 明·顾璘
                    </p>
        </div>
    ''')

app.launch(show_error=True)