import json import os import subprocess from pathlib import Path import gradio as gr import librosa import numpy as np import torch from demucs.apply import apply_model from demucs.pretrained import DEFAULT_MODEL, get_model from huggingface_hub import hf_hub_download, list_repo_files from so_vits_svc_fork.hparams import HParams from so_vits_svc_fork.inference.core import Svc ################################################################### # REPLACE THESE VALUES TO CHANGE THE MODEL REPO/CKPT NAME/SETTINGS ################################################################### # The Hugging Face Hub repo ID - 在这里修改repo_id,可替换成任何已经训练好的模型! repo_id = "kevinwang676/talktalkai-qing" # If None, Uses latest ckpt in the repo ckpt_name = None # If None, Uses "kmeans.pt" if it exists in the repo cluster_model_name = None # Set the default f0 type to use - use the one it was trained on. # The default for so-vits-svc-fork is "dio". # Options: "crepe", "crepe-tiny", "parselmouth", "dio", "harvest" default_f0_method = "crepe" # The default ratio of cluster inference to SVC inference. # If cluster_model_name is not found in the repo, this is set to 0. default_cluster_infer_ratio = 0.5 # Limit on duration of audio at inference time. increase if you can # In this parent app, we set the limit with an env var to 30 seconds # If you didnt set env var + you go OOM try changing 9e9 to <=300ish duration_limit = int(os.environ.get("MAX_DURATION_SECONDS", 9e9)) ################################################################### # Figure out the latest generator by taking highest value one. # Ex. if the repo has: G_0.pth, G_100.pth, G_200.pth, we'd use G_200.pth if ckpt_name is None: latest_id = sorted( [ int(Path(x).stem.split("_")[1]) for x in list_repo_files(repo_id) if x.startswith("G_") and x.endswith(".pth") ] )[-1] ckpt_name = f"G_{latest_id}.pth" cluster_model_name = cluster_model_name or "kmeans.pt" if cluster_model_name in list_repo_files(repo_id): print(f"Found Cluster model - Downloading {cluster_model_name} from {repo_id}") cluster_model_path = hf_hub_download(repo_id, cluster_model_name) else: print(f"Could not find {cluster_model_name} in {repo_id}. Using None") cluster_model_path = None default_cluster_infer_ratio = default_cluster_infer_ratio if cluster_model_path else 0 generator_path = hf_hub_download(repo_id, ckpt_name) config_path = hf_hub_download(repo_id, "config.json") hparams = HParams(**json.loads(Path(config_path).read_text())) speakers = list(hparams.spk.keys()) device = "cuda" if torch.cuda.is_available() else "cpu" model = Svc(net_g_path=generator_path, config_path=config_path, device=device, cluster_model_path=cluster_model_path) demucs_model = get_model(DEFAULT_MODEL) def extract_vocal_demucs(model, filename, sr=44100, device=None, shifts=1, split=True, overlap=0.25, jobs=0): wav, sr = librosa.load(filename, mono=False, sr=sr) wav = torch.tensor(wav) ref = wav.mean(0) wav = (wav - ref.mean()) / ref.std() sources = apply_model( model, wav[None], device=device, shifts=shifts, split=split, overlap=overlap, progress=True, num_workers=jobs )[0] sources = sources * ref.std() + ref.mean() # We take just the vocals stem. I know the vocals for this model are at index -1 # If using different model, check model.sources.index('vocals') vocal_wav = sources[-1] # I did this because its the same normalization the so-vits model required vocal_wav = vocal_wav / max(1.01 * vocal_wav.abs().max(), 1) vocal_wav = vocal_wav.numpy() vocal_wav = librosa.to_mono(vocal_wav) vocal_wav = vocal_wav.T instrumental_wav = sources[:-1].sum(0).numpy().T return vocal_wav, instrumental_wav def download_youtube_clip( video_identifier, start_time, end_time, output_filename, num_attempts=5, url_base="https://www.youtube.com/watch?v=", quiet=False, force=False, ): output_path = Path(output_filename) if output_path.exists(): if not force: return output_path else: output_path.unlink() quiet = "--quiet --no-warnings" if quiet else "" command = f""" yt-dlp {quiet} -x --audio-format wav -f bestaudio -o "{output_filename}" --download-sections "*{start_time}-{end_time}" "{url_base}{video_identifier}" # noqa: E501 """.strip() attempts = 0 while True: try: _ = subprocess.check_output(command, shell=True, stderr=subprocess.STDOUT) except subprocess.CalledProcessError: attempts += 1 if attempts == num_attempts: return None else: break if output_path.exists(): return output_path else: return None def predict( speaker, audio, transpose: int = 0, auto_predict_f0: bool = False, cluster_infer_ratio: float = 0, noise_scale: float = 0.4, f0_method: str = "crepe", db_thresh: int = -40, pad_seconds: float = 0.5, chunk_seconds: float = 0.5, absolute_thresh: bool = False, ): audio, _ = librosa.load(audio, sr=model.target_sample, duration=duration_limit) audio = model.infer_silence( audio.astype(np.float32), speaker=speaker, transpose=transpose, auto_predict_f0=auto_predict_f0, cluster_infer_ratio=cluster_infer_ratio, noise_scale=noise_scale, f0_method=f0_method, db_thresh=db_thresh, pad_seconds=pad_seconds, chunk_seconds=chunk_seconds, absolute_thresh=absolute_thresh, ) return model.target_sample, audio def predict_song_from_yt( ytid_or_url, start, end, speaker=speakers[0], transpose: int = 0, auto_predict_f0: bool = False, cluster_infer_ratio: float = 0, noise_scale: float = 0.4, f0_method: str = "dio", db_thresh: int = -40, pad_seconds: float = 0.5, chunk_seconds: float = 0.5, absolute_thresh: bool = False, ): end = min(start + duration_limit, end) original_track_filepath = download_youtube_clip( ytid_or_url, start, end, "track.wav", force=True, url_base="" if ytid_or_url.startswith("http") else "https://www.youtube.com/watch?v=", ) vox_wav, inst_wav = extract_vocal_demucs(demucs_model, original_track_filepath) if transpose != 0: inst_wav = librosa.effects.pitch_shift(inst_wav.T, sr=model.target_sample, n_steps=transpose).T cloned_vox = model.infer_silence( vox_wav.astype(np.float32), speaker=speaker, transpose=transpose, auto_predict_f0=auto_predict_f0, cluster_infer_ratio=cluster_infer_ratio, noise_scale=noise_scale, f0_method=f0_method, db_thresh=db_thresh, pad_seconds=pad_seconds, chunk_seconds=chunk_seconds, absolute_thresh=absolute_thresh, ) full_song = inst_wav + np.expand_dims(cloned_vox, 1) return (model.target_sample, full_song), (model.target_sample, cloned_vox) description = f""" ##
🏞️ - 滔滔AI,为您提供全场景的AI声音服务(如AI拟声、AI歌手、AI变声等)
###
🌟 - 滔滔AI合作音乐人:[一清清清](https://space.bilibili.com/22960772?spm_id_from=333.337.0.0);AI歌手,唱我想唱!
###
🎡 - 更多精彩,尽在[滔滔AI](http://www.talktalkai.com);合作:talktalkai.kevin@gmail.com
💡 - 如何使用此程序:在页面上方选择“从B站视频上传”模块,填写视频网址和视频起止时间后,点击“submit”按键即可!您还可以点击页面最下方的示例快速预览效果

talktalkai

""".strip() article = """

注意❗:请不要生成会对个人以及组织造成侵害的内容,此程序仅供科研、学习及个人娱乐使用。

🌊🎶🏞️ - 江水东流急,滔滔无尽声。 明·顾璘

""".strip() interface_mic = gr.Interface( predict, inputs=[ gr.Dropdown(speakers, value=speakers[0], label="🎤AI歌手🎶 - 🌟一清清清🌟"), gr.Audio(type="filepath", source="microphone", label="请用麦克风上传您想转换的歌曲"), gr.Slider(-12, 12, value=0, step=1, label="变调 (默认为0;有正负值,+2为升高两个key)"), gr.Checkbox(False, label="是否开启自动f0预测", info="勾选即为开启;配合聚类模型f0预测效果更好,仅限语音转换时使用", visible=False), gr.Slider(0.0, 1.0, value=default_cluster_infer_ratio, step=0.1, label="聚类模型混合比例", info="0-1之间,0即不启用聚类。使用聚类模型能提升音色相似度,但会导致咬字下降 (如果使用,建议0.5左右)"), gr.Slider(0.0, 1.0, value=0.4, step=0.1, label="noise scale (建议保持不变)", visible=False), gr.Dropdown( choices=["crepe", "crepe-tiny", "parselmouth", "dio", "harvest"], value=default_f0_method, label="模型推理方法 (crepe推理效果最好)", visible=False ), ], outputs="audio", title="🌊💕🎶 - 滔滔AI+音乐:可从B站直接上传素材,无需分离背景音", description=description, article=article, ) interface_file = gr.Interface( predict, inputs=[ gr.Dropdown(speakers, value=speakers[0], label="🎤AI歌手🎶 - 🌟一清清清🌟"), gr.Audio(type="filepath", source="upload", label="请上传您想转换的歌曲 (仅人声部分)"), gr.Slider(-12, 12, value=0, step=1, label="变调 (默认为0;有正负值,+2为升高两个key)"), gr.Checkbox(False, label="是否开启自动f0预测", info="勾选即为开启;配合聚类模型f0预测效果更好,仅限语音转换时使用", visible=False), gr.Slider(0.0, 1.0, value=default_cluster_infer_ratio, step=0.1, label="聚类模型混合比例", info="0-1之间,0即不启用聚类。使用聚类模型能提升音色相似度,但会导致咬字下降 (如果使用,建议0.5左右)"), gr.Slider(0.0, 1.0, value=0.4, step=0.1, label="noise scale (建议保持不变)", visible=False), gr.Dropdown( choices=["crepe", "crepe-tiny", "parselmouth", "dio", "harvest"], value=default_f0_method, label="模型推理方法 (crepe推理效果最好)", visible=False ), ], outputs="audio", title="🌊💕🎶 - 滔滔AI+音乐:可从B站直接上传素材,无需分离背景音", description=description, article=article, ) interface_yt = gr.Interface( predict_song_from_yt, inputs=[ gr.Textbox( label="Bilibili网址", info="请填写含有您喜欢歌曲的Bilibili网址,可直接填写相应的BV号", value="https://www.bilibili.com/video/BV..." ), gr.Number(value=0, label="起始时间 (秒)"), gr.Number(value=15, label="结束时间 (秒)"), gr.Dropdown(speakers, value=speakers[0], label="🎤AI歌手🎶 - 🌟一清清清🌟"), gr.Slider(-12, 12, value=0, step=1, label="变调 (默认为0;有正负值,+2为升高两个key)"), gr.Checkbox(False, label="是否开启自动f0预测", info="勾选即为开启;配合聚类模型f0预测效果更好,仅限语音转换时使用", visible=False), gr.Slider(0.0, 1.0, value=default_cluster_infer_ratio, step=0.1, label="聚类模型混合比例", info="0-1之间,0即不启用聚类。使用聚类模型能提升音色相似度,但会导致咬字下降"), gr.Slider(0.0, 1.0, value=0.4, step=0.1, label="noise scale (建议保持不变)", visible=False), gr.Dropdown( choices=["crepe", "crepe-tiny", "parselmouth", "dio", "harvest"], value=default_f0_method, label="模型推理方法 (crepe推理效果最好)", visible=False ), ], outputs=[gr.Audio(label="AI歌手+伴奏🎵"), gr.Audio(label="AI歌手人声部分🎤")], title="🌊💕🎶 - 滔滔AI+音乐:可从B站直接上传素材,无需分离背景音", description=description, article=article, examples=[ ["https://www.bilibili.com/video/BV1ip4y1p7Pn", 87, 103, speakers[0], 0, False, default_cluster_infer_ratio, 0.4, default_f0_method], ], ) interface = gr.TabbedInterface( [interface_yt, interface_mic, interface_file], ["📺 - 从B站视频上传 ⭐推荐⭐", "🎙️ - 从麦克风上传", "🎵 - 从文件上传"], ) if __name__ == "__main__": interface.launch(show_error=True)