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 import spaces # 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_298000_seed_uvit_facodec_small_wavenet_pruned.pth", "config_dit_mel_seed_facodec_small_wavenet.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'], map_location='cpu')) 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) speech_tokenizer_type = config['model_params']['speech_tokenizer'].get('type', 'cosyvoice') if speech_tokenizer_type == 'cosyvoice': 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) elif speech_tokenizer_type == 'facodec': ckpt_path, config_path = load_custom_model_from_hf("Plachta/FAcodec", 'pytorch_model.bin', 'config.yml') codec_config = yaml.safe_load(open(config_path)) codec_model_params = recursive_munch(codec_config['model_params']) codec_encoder = build_model(codec_model_params, stage="codec") ckpt_params = torch.load(ckpt_path, map_location="cpu") for key in codec_encoder: codec_encoder[key].load_state_dict(ckpt_params[key], strict=False) _ = [codec_encoder[key].eval() for key in codec_encoder] _ = [codec_encoder[key].to(device) for key in codec_encoder] # 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) # f0 conditioned model dit_checkpoint_path, dit_config_path = load_custom_model_from_hf("Plachta/Seed-VC", "DiT_step_404000_seed_v2_uvit_facodec_small_wavenet_f0_pruned.pth", "config_dit_mel_seed_facodec_small_wavenet_f0.yml") config = yaml.safe_load(open(dit_config_path, 'r')) model_params = recursive_munch(config['model_params']) model_f0 = build_model(model_params, stage='DiT') hop_length = config['preprocess_params']['spect_params']['hop_length'] sr = config['preprocess_params']['sr'] # Load checkpoints model_f0, _, _, _ = load_checkpoint(model_f0, None, dit_checkpoint_path, load_only_params=True, ignore_modules=[], is_distributed=False) for key in model_f0: model_f0[key].eval() model_f0[key].to(device) model_f0.cfm.estimator.setup_caches(max_batch_size=1, max_seq_length=8192) # f0 extractor from modules.rmvpe import RMVPE model_path = load_custom_model_from_hf("lj1995/VoiceConversionWebUI", "rmvpe.pt", None) rmvpe = RMVPE(model_path, is_half=False, device=device) def adjust_f0_semitones(f0_sequence, n_semitones): factor = 2 ** (n_semitones / 12) return f0_sequence * factor @spaces.GPU @torch.no_grad() @torch.inference_mode() def voice_conversion(source, target, diffusion_steps, length_adjust, inference_cfg_rate, n_quantizers): inference_module = model if not f0_condition else model_f0 # 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().to(device) ref_audio = torch.tensor(ref_audio[:sr * 30]).unsqueeze(0).float().to(device) # Resample source_waves_16k = torchaudio.functional.resample(source_audio, sr, 16000) ref_waves_16k = torchaudio.functional.resample(ref_audio, sr, 16000) # Extract features if speech_tokenizer_type == 'cosyvoice': S_alt = cosyvoice_frontend.extract_speech_token(source_waves_16k)[0] S_ori = cosyvoice_frontend.extract_speech_token(ref_waves_16k)[0] elif speech_tokenizer_type == 'facodec': converted_waves_24k = torchaudio.functional.resample(source_audio, sr, 24000) wave_lengths_24k = torch.LongTensor([converted_waves_24k.size(1)]).to(converted_waves_24k.device) waves_input = converted_waves_24k.unsqueeze(1) z = codec_encoder.encoder(waves_input) ( quantized, codes ) = codec_encoder.quantizer( z, waves_input, ) S_alt = torch.cat([codes[1], codes[0]], dim=1) # S_ori should be extracted in the same way waves_24k = torchaudio.functional.resample(ref_audio, sr, 24000) waves_input = waves_24k.unsqueeze(1) z = codec_encoder.encoder(waves_input) ( quantized, codes ) = codec_encoder.quantizer( z, waves_input, ) S_ori = torch.cat([codes[1], codes[0]], dim=1) mel = to_mel(source_audio.to(device).float()) mel2 = to_mel(ref_audio.to(device).float()) target_lengths = torch.LongTensor([int(mel.size(2) * length_adjust)]).to(mel.device) target2_lengths = torch.LongTensor([mel2.size(2)]).to(mel2.device) 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)) if f0_condition: waves_16k = torchaudio.functional.resample(waves_24k, sr, 16000) converted_waves_16k = torchaudio.functional.resample(converted_waves_24k, sr, 16000) F0_ori = rmvpe.infer_from_audio(waves_16k[0], thred=0.03) F0_alt = rmvpe.infer_from_audio(converted_waves_16k[0], thred=0.03) F0_ori = torch.from_numpy(F0_ori).to(device)[None] F0_alt = torch.from_numpy(F0_alt).to(device)[None] voiced_F0_ori = F0_ori[F0_ori > 1] voiced_F0_alt = F0_alt[F0_alt > 1] log_f0_alt = torch.log(F0_alt + 1e-5) voiced_log_f0_ori = torch.log(voiced_F0_ori + 1e-5) voiced_log_f0_alt = torch.log(voiced_F0_alt + 1e-5) median_log_f0_ori = torch.median(voiced_log_f0_ori) median_log_f0_alt = torch.median(voiced_log_f0_alt) # mean_log_f0_ori = torch.mean(voiced_log_f0_ori) # mean_log_f0_alt = torch.mean(voiced_log_f0_alt) # shift alt log f0 level to ori log f0 level shifted_log_f0_alt = log_f0_alt.clone() if auto_f0_adjust: shifted_log_f0_alt[F0_alt > 1] = log_f0_alt[F0_alt > 1] - median_log_f0_alt + median_log_f0_ori shifted_f0_alt = torch.exp(shifted_log_f0_alt) if pitch_shift != 0: shifted_f0_alt[F0_alt > 1] = adjust_f0_semitones(shifted_f0_alt[F0_alt > 1], pitch_shift) else: F0_ori = None F0_alt = None shifted_f0_alt = None # Length regulation cond = inference_module.length_regulator(S_alt, ylens=target_lengths, n_quantizers=int(n_quantizers), f0=shifted_f0_alt)[0] prompt_condition = inference_module.length_regulator(S_ori, ylens=target2_lengths, n_quantizers=int(n_quantizers), f0=F0_ori)[0] cat_condition = torch.cat([prompt_condition, cond], dim=1) # Voice Conversion vc_target = inference_module.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 # if f0_condition: # f04vocoder = torch.nn.functional.interpolate(shifted_f0_alt.unsqueeze(1), size=vc_target.size(-1), # mode='nearest').squeeze(1) # else: f04vocoder = None vc_wave = hift_gen.inference(vc_target, f0=f04vocoder) 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=200, value=10, step=1, label="Diffusion Steps", info="10 by default, 50~100 for best quality"), gr.Slider(minimum=0.5, maximum=2.0, step=0.1, value=1.0, label="Length Adjust", info="<1.0 for speed-up speech, >1.0 for slow-down speech"), gr.Slider(minimum=0.0, maximum=1.0, step=0.1, value=0.7, label="Inference CFG Rate", info="has subtle influence"), gr.Slider(minimum=1, maximum=3, step=1, value=3, label="N Quantizers", info="the less quantizer used, the less prosody of source audio is preserved"), gr.Checkbox(label="Use F0 conditioned model", value=False, info="Must set to true for singing voice conversion"), gr.Checkbox(label="Auto F0 adjust", value=True, info="Roughly adjust F0 to match target voice. Only works when F0 conditioned model is used."), gr.Slider(label='Pitch shift', minimum=-24, maximum=24, step=1, value=0, info='Pitch shift in semitones, only works when F0 conditioned model is used'), ] examples = [["examples/source/yae_0.wav", "examples/reference/dingzhen_0.wav", 50, 1.0, 0.7, 1, False, True, 0],] outputs = gr.Audio(label="Output Audio") gr.Interface(fn=voice_conversion, description=description, inputs=inputs, outputs=outputs, title="Seed Voice Conversion", examples=examples, cache_examples=False, ).launch()