import spaces import gradio as gr import torch import torchaudio import librosa from modules.commons import build_model, load_checkpoint, recursive_munch import yaml from hf_utils import load_custom_model_from_hf # Load model and configuration device = torch.device("cuda" if torch.cuda.is_available() else "cpu") dit_checkpoint_path, dit_config_path = load_custom_model_from_hf("Plachta/Seed-VC", "DiT_step_315000_seed_v2_online_pruned.pth", "config_dit_mel_seed.yml") config = yaml.safe_load(open(dit_config_path, 'r')) model_params = recursive_munch(config['model_params']) model = build_model(model_params, stage='DiT') hop_length = config['preprocess_params']['spect_params']['hop_length'] sr = config['preprocess_params']['sr'] # Load checkpoints model, _, _, _ = load_checkpoint(model, None, dit_checkpoint_path, load_only_params=True, ignore_modules=[], is_distributed=False) for key in model: model[key].eval() model[key].to(device) model.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192) # Load additional modules from modules.campplus.DTDNN import CAMPPlus campplus_model = CAMPPlus(feat_dim=80, embedding_size=192) campplus_model.load_state_dict(torch.load(config['model_params']['style_encoder']['campplus_path'])) campplus_model.eval() campplus_model.to(device) from modules.hifigan.generator import HiFTGenerator from modules.hifigan.f0_predictor import ConvRNNF0Predictor hift_checkpoint_path, hift_config_path = load_custom_model_from_hf("Plachta/Seed-VC", "hift.pt", "hifigan.yml") hift_config = yaml.safe_load(open(hift_config_path, 'r')) hift_gen = HiFTGenerator(**hift_config['hift'], f0_predictor=ConvRNNF0Predictor(**hift_config['f0_predictor'])) hift_gen.load_state_dict(torch.load(hift_checkpoint_path, map_location='cpu')) hift_gen.eval() hift_gen.to(device) from modules.cosyvoice_tokenizer.frontend import CosyVoiceFrontEnd speech_tokenizer_path = load_custom_model_from_hf("Plachta/Seed-VC", "speech_tokenizer_v1.onnx", None) cosyvoice_frontend = CosyVoiceFrontEnd(speech_tokenizer_model=speech_tokenizer_path, device='cuda', device_id=0) # Generate mel spectrograms mel_fn_args = { "n_fft": config['preprocess_params']['spect_params']['n_fft'], "win_size": config['preprocess_params']['spect_params']['win_length'], "hop_size": config['preprocess_params']['spect_params']['hop_length'], "num_mels": config['preprocess_params']['spect_params']['n_mels'], "sampling_rate": sr, "fmin": 0, "fmax": 8000, "center": False } from modules.audio import mel_spectrogram to_mel = lambda x: mel_spectrogram(x, **mel_fn_args) @spaces.GPU @torch.no_grad() def voice_conversion(source, target, diffusion_steps, length_adjust, inference_cfg_rate): # Load audio source_audio = librosa.load(source, sr=sr)[0] ref_audio = librosa.load(target, sr=sr)[0] # Process audio source_audio = torch.tensor(source_audio[:sr * 30]).unsqueeze(0).float() ref_audio = torch.tensor(ref_audio[:sr * 30]).unsqueeze(0).float() # Resample source_waves_16k = torchaudio.functional.resample(source_audio, sr, 16000) ref_waves_16k = torchaudio.functional.resample(ref_audio, sr, 16000) # Extract features S_alt = cosyvoice_frontend.extract_speech_token(source_waves_16k)[0] S_ori = cosyvoice_frontend.extract_speech_token(ref_waves_16k)[0] mel = to_mel(source_audio.float()) mel2 = to_mel(ref_audio.float()) target_lengths = torch.LongTensor([int(mel.size(2) * length_adjust)]).to(mel.device) target2_lengths = torch.LongTensor([mel2.size(2)]).to(mel2.device) # Style encoding feat = torchaudio.compliance.kaldi.fbank(source_waves_16k, num_mel_bins=80, dither=0, sample_frequency=16000) feat = feat - feat.mean(dim=0, keepdim=True) style1 = campplus_model(feat.unsqueeze(0)) feat2 = torchaudio.compliance.kaldi.fbank(ref_waves_16k, num_mel_bins=80, dither=0, sample_frequency=16000) feat2 = feat2 - feat2.mean(dim=0, keepdim=True) style2 = campplus_model(feat2.unsqueeze(0)) # Length regulation cond = model.length_regulator(S_alt, ylens=target_lengths)[0] prompt_condition = model.length_regulator(S_ori, ylens=target2_lengths)[0] cat_condition = torch.cat([prompt_condition, cond], dim=1) # Voice Conversion vc_target = model.cfm.inference(cat_condition, torch.LongTensor([cat_condition.size(1)]).to(mel2.device), mel2, style2, None, diffusion_steps, inference_cfg_rate=inference_cfg_rate) vc_target = vc_target[:, :, mel2.size(-1):] # Convert to waveform vc_wave = hift_gen.inference(vc_target) return (sr, vc_wave.squeeze(0).cpu().numpy()) if __name__ == "__main__": description = "Zero-shot voice conversion with in-context learning. Check out our [GitHub repository](https://github.com/Plachtaa/seed-vc) for details and updates." inputs = [ gr.Audio(type="filepath", label="Source Audio"), gr.Audio(type="filepath", label="Reference Audio"), gr.Slider(minimum=1, maximum=1000, value=100, step=1, label="Diffusion Steps"), gr.Slider(minimum=0.5, maximum=2.0, step=0.1, value=1.0, label="Length Adjust"), gr.Slider(minimum=0.1, maximum=1.0, step=0.1, value=0.7, label="Inference CFG Rate"), ] outputs = gr.Audio(label="Output Audio") gr.Interface(fn=voice_conversion, description=description, inputs=inputs, outputs=outputs, title="Seed Voice Conversion").launch()