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| import os | |
| import glob | |
| import json | |
| import traceback | |
| import logging | |
| import gradio as gr | |
| import numpy as np | |
| import librosa | |
| import torch | |
| import asyncio | |
| import edge_tts | |
| import yt_dlp | |
| import ffmpeg | |
| import subprocess | |
| import sys | |
| import io | |
| import wave | |
| import gc | |
| from datetime import datetime | |
| from concurrent.futures import ThreadPoolExecutor, ProcessPoolExecutor | |
| from functools import lru_cache | |
| import psutil | |
| from fairseq import checkpoint_utils | |
| from lib.infer_pack.models import ( | |
| SynthesizerTrnMs256NSFsid, | |
| SynthesizerTrnMs256NSFsid_nono, | |
| SynthesizerTrnMs768NSFsid, | |
| SynthesizerTrnMs768NSFsid_nono, | |
| ) | |
| from vc_infer_pipeline import VC | |
| from config import Config | |
| config = Config() | |
| logging.getLogger("numba").setLevel(logging.WARNING) | |
| limitation = os.getenv("SYSTEM") == "spaces" | |
| # ============================== | |
| # OPTIMIZATION SETTINGS | |
| # ============================== | |
| MAX_WORKERS = 1 # Karena hanya 1 core | |
| CHUNK_SIZE = 10 * 16000 # 10 detik pada 16kHz (optimal untuk single core) | |
| OVERLAP_SIZE = int(0.5 * 16000) # 0.5 detik overlap untuk menghindari artifacts | |
| MEMORY_THRESHOLD = 85 # Clear cache jika memory usage > 85% | |
| USE_HALF_PRECISION = True # Gunakan float16 jika memungkinkan | |
| CACHE_MODELS = True # Cache model yang sudah dimuat | |
| # ============================== | |
| # MEMORY MANAGEMENT | |
| # ============================== | |
| class MemoryManager: | |
| def check_memory(): | |
| """Cek penggunaan memori""" | |
| memory = psutil.virtual_memory() | |
| return memory.percent | |
| def clear_cache(): | |
| """Bersihkan cache memori""" | |
| if torch.cuda.is_available(): | |
| torch.cuda.empty_cache() | |
| gc.collect() | |
| def auto_clear(): | |
| """Auto-clear jika memori tinggi""" | |
| if MemoryManager.check_memory() > MEMORY_THRESHOLD: | |
| print(f"Memory high ({MemoryManager.check_memory()}%), clearing cache...") | |
| MemoryManager.clear_cache() | |
| # ============================== | |
| # CHUNK PROCESSING SYSTEM | |
| # ============================== | |
| class ChunkProcessor: | |
| def __init__(self, vc_pipeline, chunk_size=CHUNK_SIZE, overlap=OVERLAP_SIZE): | |
| self.vc_pipeline = vc_pipeline | |
| self.chunk_size = chunk_size | |
| self.overlap = overlap | |
| def split_into_chunks(self, audio): | |
| """Split audio menjadi chunks dengan overlap""" | |
| audio_len = len(audio) | |
| chunks = [] | |
| if audio_len <= self.chunk_size: | |
| return [audio], [0] | |
| start = 0 | |
| while start < audio_len: | |
| end = min(start + self.chunk_size, audio_len) | |
| chunk = audio[start:end] | |
| chunks.append(chunk) | |
| start += self.chunk_size - self.overlap | |
| return chunks, list(range(0, audio_len, self.chunk_size - self.overlap)) | |
| def process_chunk(self, chunk_data): | |
| """Process single chunk""" | |
| chunk, index, params = chunk_data | |
| try: | |
| # Ekstrak parameters | |
| (hubert_model, net_g, sid, vc_input, times, f0_up_key, | |
| f0_method, file_index, index_rate, if_f0, filter_radius, | |
| tgt_sr, resample_sr, rms_mix_rate, version, protect) = params | |
| # Process chunk | |
| chunk_audio = self.vc_pipeline( | |
| hubert_model, | |
| net_g, | |
| sid, | |
| chunk, | |
| vc_input, | |
| times, | |
| f0_up_key, | |
| f0_method, | |
| file_index, | |
| index_rate, | |
| if_f0, | |
| filter_radius, | |
| tgt_sr, | |
| resample_sr, | |
| rms_mix_rate, | |
| version, | |
| protect, | |
| f0_file=None, | |
| ) | |
| return index, chunk_audio | |
| except Exception as e: | |
| print(f"Error processing chunk {index}: {e}") | |
| return index, None | |
| def process_parallel(self, audio, params, max_workers=MAX_WORKERS): | |
| """Process chunks dengan parallel execution""" | |
| chunks, indices = self.split_into_chunks(audio) | |
| if len(chunks) == 1: | |
| # Direct processing untuk audio pendek | |
| return self.vc_pipeline(params[0], params[1], params[2], audio, | |
| *params[3:]) | |
| # Prepare chunk data | |
| chunk_data = [(chunk, idx, params) for chunk, idx in zip(chunks, indices)] | |
| # Sequential processing untuk 1 core (lebih stabil) | |
| results = [] | |
| for data in chunk_data: | |
| idx, result = self.process_chunk(data) | |
| if result is not None: | |
| results.append((idx, result)) | |
| # Sort by original index | |
| results.sort(key=lambda x: x[0]) | |
| # Merge chunks with crossfade | |
| return self.merge_chunks([r[1] for r in results]) | |
| def merge_chunks(self, chunks): | |
| """Merge chunks dengan crossfade untuk menghindari artifacts""" | |
| if not chunks: | |
| return np.array([], dtype=np.float32) | |
| if len(chunks) == 1: | |
| return chunks[0] | |
| merged = chunks[0] | |
| for i in range(1, len(chunks)): | |
| current_chunk = chunks[i] | |
| if len(merged) < self.overlap or len(current_chunk) < self.overlap: | |
| # Jika chunk terlalu pendek, langsung concatenate | |
| merged = np.concatenate([merged, current_chunk]) | |
| else: | |
| # Crossfade overlap region | |
| fade_out = merged[-self.overlap:] | |
| fade_in = current_chunk[:self.overlap] | |
| # Linear crossfade | |
| t = np.linspace(0, 1, self.overlap) | |
| faded = fade_out * (1 - t) + fade_in * t | |
| # Merge | |
| merged = np.concatenate([ | |
| merged[:-self.overlap], | |
| faded, | |
| current_chunk[self.overlap:] | |
| ]) | |
| return merged | |
| # ============================== | |
| # MODEL CACHE SYSTEM | |
| # ============================== | |
| class ModelCache: | |
| _instance = None | |
| _models = {} | |
| _hubert = None | |
| def __new__(cls): | |
| if cls._instance is None: | |
| cls._instance = super(ModelCache, cls).__new__(cls) | |
| return cls._instance | |
| def get_hubert(cls): | |
| if cls._hubert is None: | |
| cls._hubert = cls.load_hubert() | |
| return cls._hubert | |
| def load_hubert(cls): | |
| """Load hubert model dengan optimasi""" | |
| print("Loading HuBERT model...") | |
| models, _, _ = checkpoint_utils.load_model_ensemble_and_task( | |
| ["hubert_base.pt"], | |
| suffix="", | |
| ) | |
| hubert_model = models[0] | |
| hubert_model = hubert_model.to(config.device) | |
| if USE_HALF_PRECISION and config.is_half: | |
| hubert_model = hubert_model.half() | |
| else: | |
| hubert_model = hubert_model.float() | |
| hubert_model.eval() | |
| return hubert_model | |
| def get_model(cls, model_path): | |
| """Get model from cache atau load baru""" | |
| if model_path in cls._models and CACHE_MODELS: | |
| print(f"Using cached model: {model_path}") | |
| return cls._models[model_path] | |
| return None | |
| def cache_model(cls, model_path, model_data): | |
| """Cache model""" | |
| if CACHE_MODELS: | |
| cls._models[model_path] = model_data | |
| def clear_model_cache(cls): | |
| """Clear model cache""" | |
| cls._models.clear() | |
| if torch.cuda.is_available(): | |
| torch.cuda.empty_cache() | |
| gc.collect() | |
| # ============================== | |
| # OPTIMIZED VC FUNCTION | |
| # ============================== | |
| def create_vc_fn(model_title, tgt_sr, net_g, vc, if_f0, version, file_index): | |
| def vc_fn( | |
| vc_audio_mode, | |
| vc_input, | |
| vc_upload, | |
| tts_text, | |
| tts_voice, | |
| f0_up_key, | |
| f0_method, | |
| index_rate, | |
| filter_radius, | |
| resample_sr, | |
| rms_mix_rate, | |
| protect, | |
| ): | |
| try: | |
| # Auto clear memory sebelum proses | |
| MemoryManager.auto_clear() | |
| # Load audio | |
| if vc_audio_mode == "Input path" or "Youtube" and vc_input != "": | |
| # Gunakan librosa dengan optimasi | |
| audio, sr = librosa.load(vc_input, sr=16000, mono=True) | |
| elif vc_audio_mode == "Upload audio": | |
| if vc_upload is None: | |
| return "You need to upload an audio", None | |
| sampling_rate, audio = vc_upload | |
| # Batasi durasi jika di space | |
| if limitation: | |
| duration = audio.shape[0] / sampling_rate | |
| if duration > 360: | |
| return "Please upload an audio file that is less than 1 minute.", None | |
| # Konversi audio | |
| audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32) | |
| if len(audio.shape) > 1: | |
| audio = librosa.to_mono(audio.transpose(1, 0)) | |
| if sampling_rate != 16000: | |
| audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000) | |
| elif vc_audio_mode == "TTS Audio": | |
| if limitation and len(tts_text) > 600: | |
| return "Text is too long", None | |
| if tts_text is None or tts_voice is None: | |
| return "You need to enter text and select a voice", None | |
| # Generate TTS | |
| asyncio.run(edge_tts.Communicate(tts_text, "-".join(tts_voice.split('-')[:-1])).save("tts.mp3")) | |
| audio, sr = librosa.load("tts.mp3", sr=16000, mono=True) | |
| vc_input = "tts.mp3" | |
| else: | |
| return "Invalid audio mode", None | |
| # Persiapkan parameters | |
| hubert_model = ModelCache.get_hubert() | |
| times = [0, 0, 0] | |
| f0_up_key = int(f0_up_key) | |
| # Parameters untuk chunk processing | |
| params = ( | |
| hubert_model, | |
| net_g, | |
| 0, | |
| vc_input, | |
| times, | |
| f0_up_key, | |
| f0_method, | |
| file_index, | |
| index_rate, | |
| if_f0, | |
| filter_radius, | |
| tgt_sr, | |
| resample_sr, | |
| rms_mix_rate, | |
| version, | |
| protect | |
| ) | |
| # Gunakan chunk processor untuk audio panjang | |
| if len(audio) > CHUNK_SIZE * 2: # Hanya chunk jika > 20 detik | |
| print(f"Processing {len(audio)/16000:.2f}s audio in chunks...") | |
| processor = ChunkProcessor(vc.pipeline) | |
| audio_opt = processor.process_parallel(audio, params, max_workers=MAX_WORKERS) | |
| else: | |
| # Direct processing untuk audio pendek | |
| audio_opt = vc.pipeline( | |
| hubert_model, | |
| net_g, | |
| 0, | |
| audio, | |
| vc_input, | |
| times, | |
| f0_up_key, | |
| f0_method, | |
| file_index, | |
| index_rate, | |
| if_f0, | |
| filter_radius, | |
| tgt_sr, | |
| resample_sr, | |
| rms_mix_rate, | |
| version, | |
| protect, | |
| f0_file=None, | |
| ) | |
| info = f"[{datetime.now().strftime('%Y-%m-%d %H:%M')}]: npy: {times[0]:.2f}s, f0: {times[1]:.2f}s, infer: {times[2]:.2f}s" | |
| print(f"{model_title} | {info}") | |
| # Clear memory setelah proses | |
| MemoryManager.auto_clear() | |
| return info, (tgt_sr, audio_opt) | |
| except Exception as e: | |
| info = f"Error: {str(e)}\n{traceback.format_exc()}" | |
| print(info) | |
| MemoryManager.auto_clear() | |
| return info, (None, None) | |
| return vc_fn | |
| # ============================== | |
| # OPTIMIZED MODEL LOADING | |
| # ============================== | |
| def load_model(): | |
| categories = [] | |
| with open("weights/folder_info.json", "r", encoding="utf-8") as f: | |
| folder_info = json.load(f) | |
| for category_name, category_info in folder_info.items(): | |
| if not category_info['enable']: | |
| continue | |
| category_title = category_info['title'] | |
| category_folder = category_info['folder_path'] | |
| models = [] | |
| with open(f"weights/{category_folder}/model_info.json", "r", encoding="utf-8") as f: | |
| models_info = json.load(f) | |
| for character_name, info in models_info.items(): | |
| if not info['enable']: | |
| continue | |
| model_title = info['title'] | |
| model_name = info['model_path'] | |
| model_author = info.get("author", None) | |
| model_cover = f"weights/{category_folder}/{character_name}/{info['cover']}" | |
| model_index = f"weights/{category_folder}/{character_name}/{info['feature_retrieval_library']}" | |
| # Cek cache dulu | |
| model_path = f"weights/{category_folder}/{character_name}/{model_name}" | |
| cached_model = ModelCache.get_model(model_path) | |
| if cached_model: | |
| cpt, tgt_sr, if_f0, version, net_g = cached_model | |
| else: | |
| # Load model dengan memory optimasi | |
| cpt = torch.load(model_path, map_location="cpu", weights_only=False) | |
| tgt_sr = cpt["config"][-1] | |
| cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk | |
| if_f0 = cpt.get("f0", 1) | |
| version = cpt.get("version", "v1") | |
| # Load model architecture | |
| if version == "v1": | |
| if if_f0 == 1: | |
| net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half) | |
| else: | |
| net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"]) | |
| elif version == "v2": | |
| if if_f0 == 1: | |
| net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half) | |
| else: | |
| net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"]) | |
| # Load weights | |
| del net_g.enc_q | |
| net_g.load_state_dict(cpt["weight"], strict=False) | |
| net_g.eval().to(config.device) | |
| if USE_HALF_PRECISION and config.is_half: | |
| net_g = net_g.half() | |
| else: | |
| net_g = net_g.float() | |
| # Cache model | |
| ModelCache.cache_model(model_path, (cpt, tgt_sr, if_f0, version, net_g)) | |
| # Create VC | |
| vc = VC(tgt_sr, config) | |
| models.append((character_name, model_title, model_author, model_cover, | |
| version.upper(), create_vc_fn(model_title, tgt_sr, net_g, vc, if_f0, version, model_index))) | |
| print(f"Model loaded: {character_name} | {version.upper()}") | |
| categories.append([category_title, category_folder, models]) | |
| return categories | |
| # ============================== | |
| # OPTIMIZED FUNCTIONS | |
| # ============================== | |
| def cut_vocal_and_inst(url, audio_provider, split_model): | |
| """Optimized audio splitting dengan progress feedback""" | |
| if not url: | |
| raise gr.Error("URL Required!") | |
| if not os.path.exists("dl_audio"): | |
| os.mkdir("dl_audio") | |
| try: | |
| if audio_provider == "Youtube": | |
| ydl_opts = { | |
| 'format': 'bestaudio/best', | |
| 'postprocessors': [{ | |
| 'key': 'FFmpegExtractAudio', | |
| 'preferredcodec': 'wav', | |
| }], | |
| "outtmpl": 'dl_audio/youtube_audio', | |
| 'quiet': True, | |
| 'no_warnings': True, | |
| } | |
| with yt_dlp.YoutubeDL(ydl_opts) as ydl: | |
| ydl.download([url]) | |
| audio_path = "dl_audio/youtube_audio.wav" | |
| # Optimize demucs command | |
| model_name = "htdemucs" if split_model == "htdemucs" else "mdx_extra_q" | |
| output_dir = f"output/{model_name}" | |
| # Gunakan subprocess dengan Popen untuk better control | |
| cmd = ["demucs", "--two-stems=vocals", "-n", model_name, | |
| audio_path, "-o", "output", "--quiet"] | |
| process = subprocess.Popen(cmd, stdout=subprocess.PIPE, | |
| stderr=subprocess.PIPE, text=True) | |
| # Monitor progress | |
| for line in process.stdout: | |
| if "Progress" in line: | |
| print(line.strip()) | |
| process.wait() | |
| vocal_path = f"{output_dir}/youtube_audio/vocals.wav" | |
| inst_path = f"{output_dir}/youtube_audio/no_vocals.wav" | |
| return vocal_path, inst_path, audio_path, vocal_path | |
| except Exception as e: | |
| raise gr.Error(f"Error processing audio: {str(e)}") | |
| def combine_vocal_and_inst(audio_data, audio_volume, split_model): | |
| """Optimized audio combining""" | |
| if not os.path.exists("output/result"): | |
| os.makedirs("output/result") | |
| vocal_path = "output/result/output.wav" | |
| output_path = "output/result/combine.mp3" | |
| model_name = "htdemucs" if split_model == "htdemucs" else "mdx_extra_q" | |
| inst_path = f"output/{model_name}/youtube_audio/no_vocals.wav" | |
| # Write vocal file | |
| with wave.open(vocal_path, "w") as wave_file: | |
| wave_file.setnchannels(1) | |
| wave_file.setsampwidth(2) | |
| wave_file.setframerate(audio_data[0]) | |
| wave_file.writeframes(audio_data[1].tobytes()) | |
| # Optimize ffmpeg command | |
| cmd = [ | |
| 'ffmpeg', '-y', | |
| '-i', inst_path, | |
| '-i', vocal_path, | |
| '-filter_complex', f'[1:a]volume={audio_volume}dB[v];[0:a][v]amix=inputs=2:duration=longest', | |
| '-b:a', '320k', | |
| '-c:a', 'libmp3lame', | |
| '-threads', '1', # Gunakan single thread | |
| '-loglevel', 'error', | |
| output_path | |
| ] | |
| subprocess.run(cmd, stdout=subprocess.PIPE, stderr=subprocess.PIPE) | |
| return output_path | |
| # ============================== | |
| # MAIN APPLICATION | |
| # ============================== | |
| if __name__ == '__main__': | |
| # Setup audio mode | |
| audio_mode = [] | |
| f0method_mode = [] | |
| f0method_info = "" | |
| if limitation is True: | |
| audio_mode = ["Upload audio", "TTS Audio"] | |
| f0method_mode = ["pm", "crepe", "harvest"] | |
| f0method_info = "PM is fast, rmvpe is middle, Crepe or harvest is good but it was extremely slow (Default: PM)" | |
| else: | |
| audio_mode = ["Upload audio", "Youtube", "TTS Audio"] | |
| f0method_mode = ["pm", "crepe", "harvest"] | |
| f0method_info = "PM is fast, rmvpe is middle. Crepe or harvest is good but it was extremely slow (Default: PM)" | |
| if os.path.isfile("rmvpe.pt"): | |
| f0method_mode.insert(2, "rmvpe") | |
| # Pre-load hubert model | |
| print("Initializing HuBERT model...") | |
| ModelCache.get_hubert() | |
| # Load models | |
| print("Loading RVC models...") | |
| categories = load_model() | |
| # Get TTS voices | |
| tts_voice_list = asyncio.get_event_loop().run_until_complete(edge_tts.list_voices()) | |
| voices = [f"{v['ShortName']}-{v['Gender']}" for v in tts_voice_list] | |
| # Create Gradio app | |
| with gr.Blocks(theme=gr.themes.Base(), css=""" | |
| .gradio-container {max-width: 1200px !important;} | |
| .tab-nav {scroll-behavior: smooth;} | |
| """) as app: | |
| gr.Markdown(""" | |
| # ▶️ RVC Youtuber Indonesia 👳🏿♂️ | |
| [⚡] **Optimized Version** - Chunk Processing Enabled | |
| """) | |
| for (folder_title, folder, models) in categories: | |
| with gr.TabItem(folder_title): | |
| if not models: | |
| gr.Markdown("# <center> No Model Loaded.") | |
| continue | |
| for (name, title, author, cover, model_version, vc_fn) in models: | |
| with gr.TabItem(name): | |
| with gr.Row(): | |
| gr.Markdown(f""" | |
| <div align="center"> | |
| <div><b>{title}</b></div> | |
| <div>RVC {model_version} Model</div> | |
| {f'<div>Model author: {author}</div>' if author else ""} | |
| {f'<img style="width:auto;height:300px;" src="file/{cover}">' if cover else ""} | |
| </div> | |
| """) | |
| with gr.Row(): | |
| with gr.Column(): | |
| vc_audio_mode = gr.Dropdown( | |
| label="Input voice", | |
| choices=audio_mode, | |
| value="Upload audio" | |
| ) | |
| # Audio inputs | |
| vc_input = gr.Textbox(label="Input audio path", visible=False) | |
| vc_upload = gr.Audio(label="Upload audio file", visible=True) | |
| # Youtube | |
| vc_link = gr.Textbox(label="Youtube URL", visible=False) | |
| vc_split_model = gr.Dropdown( | |
| label="Splitter Model", | |
| choices=["htdemucs", "mdx_extra_q"], | |
| value="htdemucs", | |
| visible=False | |
| ) | |
| vc_split = gr.Button("Split Audio", variant="primary", visible=False) | |
| vc_vocal_preview = gr.Audio(label="Vocal Preview", visible=False) | |
| vc_inst_preview = gr.Audio(label="Instrumental Preview", visible=False) | |
| vc_audio_preview = gr.Audio(label="Audio Preview", visible=False) | |
| # TTS | |
| tts_text = gr.Textbox(visible=False, label="TTS text") | |
| tts_voice = gr.Dropdown( | |
| label="Edge-tts speaker", | |
| choices=voices, | |
| visible=False, | |
| value="en-US-AnaNeural-Female" | |
| ) | |
| with gr.Column(): | |
| vc_transform0 = gr.Number( | |
| label="Transpose", | |
| value=0, | |
| info='Type "12" for male to female, "-12" for female to male' | |
| ) | |
| f0method0 = gr.Radio( | |
| label="Pitch extraction algorithm", | |
| info=f0method_info, | |
| choices=f0method_mode, | |
| value="pm" | |
| ) | |
| index_rate1 = gr.Slider( | |
| minimum=0, maximum=1, | |
| label="Retrieval feature ratio", | |
| value=0.4, | |
| info="Too high = robotic" | |
| ) | |
| filter_radius0 = gr.Slider( | |
| minimum=0, maximum=7, | |
| label="Median Filtering", | |
| value=1, step=1, | |
| info="Reduce breathiness" | |
| ) | |
| resample_sr0 = gr.Slider( | |
| minimum=0, maximum=48000, | |
| label="Resample output", | |
| value=0, step=1, | |
| info="0 for no resampling" | |
| ) | |
| rms_mix_rate0 = gr.Slider( | |
| minimum=0, maximum=1, | |
| label="Volume Envelope", | |
| value=1, | |
| info="1 = use output envelope" | |
| ) | |
| protect0 = gr.Slider( | |
| minimum=0, maximum=0.5, | |
| label="Voice Protection", | |
| value=0.23, step=0.01, | |
| info="Protect voiceless sounds" | |
| ) | |
| with gr.Column(): | |
| vc_log = gr.Textbox(label="Output Information") | |
| vc_output = gr.Audio(label="Output Audio") | |
| vc_convert = gr.Button("Convert", variant="primary") | |
| vc_volume = gr.Slider( | |
| minimum=0, maximum=10, | |
| label="Vocal volume", value=4, step=1, | |
| visible=False | |
| ) | |
| vc_combined_output = gr.Audio( | |
| label="Output Combined Audio", | |
| visible=False | |
| ) | |
| vc_combine = gr.Button("Combine", variant="primary", visible=False) | |
| # Connect events | |
| vc_convert.click( | |
| fn=vc_fn, | |
| inputs=[ | |
| vc_audio_mode, vc_input, vc_upload, | |
| tts_text, tts_voice, vc_transform0, | |
| f0method0, index_rate1, filter_radius0, | |
| resample_sr0, rms_mix_rate0, protect0, | |
| ], | |
| outputs=[vc_log, vc_output] | |
| ).then( | |
| fn=lambda: MemoryManager.auto_clear(), | |
| outputs=[] | |
| ) | |
| vc_split.click( | |
| fn=cut_vocal_and_inst, | |
| inputs=[vc_link, gr.Dropdown(value="Youtube", visible=False), vc_split_model], | |
| outputs=[vc_vocal_preview, vc_inst_preview, vc_audio_preview, vc_input] | |
| ) | |
| vc_combine.click( | |
| fn=combine_vocal_and_inst, | |
| inputs=[vc_output, vc_volume, vc_split_model], | |
| outputs=[vc_combined_output] | |
| ) | |
| vc_audio_mode.change( | |
| fn=lambda mode: ( | |
| gr.Textbox.update(visible=(mode == "Input path")), | |
| gr.Audio.update(visible=(mode == "Upload audio")), | |
| gr.Textbox.update(visible=(mode == "Youtube")), | |
| gr.Dropdown.update(visible=(mode == "Youtube")), | |
| gr.Button.update(visible=(mode == "Youtube")), | |
| gr.Audio.update(visible=(mode == "Youtube")), | |
| gr.Audio.update(visible=(mode == "Youtube")), | |
| gr.Audio.update(visible=(mode == "Youtube")), | |
| gr.Slider.update(visible=(mode == "Youtube")), | |
| gr.Audio.update(visible=(mode == "Youtube")), | |
| gr.Button.update(visible=(mode == "Youtube")), | |
| gr.Textbox.update(visible=(mode == "TTS Audio")), | |
| gr.Dropdown.update(visible=(mode == "TTS Audio")) | |
| ), | |
| inputs=[vc_audio_mode], | |
| outputs=[ | |
| vc_input, vc_upload, vc_link, vc_split_model, | |
| vc_split, vc_vocal_preview, vc_inst_preview, | |
| vc_audio_preview, vc_volume, vc_combined_output, | |
| vc_combine, tts_text, tts_voice | |
| ] | |
| ) | |
| # Launch app | |
| # Definisikan parameter launch | |
| launch_kwargs = { | |
| 'enable_queue': True, | |
| 'max_threads': 1, | |
| 'share': config.colab if limitation else True, | |
| 'server_name': "0.0.0.0" | |
| } | |
| app.launch(**launch_kwargs) |