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) def convert(model, src, tgt): 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 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("out.wav", hps.data.sampling_rate, audio) else: write("out.wav", 24000, audio) out = "out.wav" return out model = gr.Dropdown(choices=["FreeVC", "FreeVC-s", "FreeVC (24kHz)"], value="FreeVC",type="value", label="Model") audio1 = gr.Audio(label="Source Audio", type='filepath') audio2 = gr.Audio(label="Reference Audio", type='filepath') inputs = [model, audio1, audio2] outputs = gr.Audio(label="Output Audio", type='filepath') title = "FreeVC" description = "Gradio Demo for FreeVC: Towards High-Quality Text-Free One-Shot Voice Conversion. To use it, simply upload your audio, or click the example to load. Read more at the links below. Note: It seems that the WavLM checkpoint in HuggingFace is a little different from the one used to train FreeVC, which may degrade the performance a bit. In addition, speaker similarity can be largely affected if there are too much silence in the reference audio, so please trim it before submitting." article = "
" examples=[["FreeVC", 'p225_001.wav', 'p226_002.wav'], ["FreeVC-s", 'p226_002.wav', 'p225_001.wav'], ["FreeVC (24kHz)", 'p225_001.wav', 'p226_002.wav']] gr.Interface(convert, inputs, outputs, title=title, description=description, article=article, examples=examples, enable_queue=True).launch()