dreamvoice/train_utils/src/freevc_wrapper.py

import os
import torch
import librosa
import soundfile as sf
from pathlib import Path

from transformers import WavLMModel
from freevc.utils import load_checkpoint, get_hparams_from_file
from freevc.models import SynthesizerTrn
# from mel_processing import mel_spectrogram_torch
# from free_vc.speaker_encoder.voice_encoder import SpeakerEncoder
from speaker_encoder.voice_encoder import SpeakerEncoder


def get_freevc_models(path='freevc', speaker_path='../pre_ckpts/spk_encoder/pretrained.pt', device='cuda'):
    hps = get_hparams_from_file("freevc/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()
    load_checkpoint("../prepare_freevc/ckpts/freevc.pth", freevc, None)

    cmodel = WavLMModel.from_pretrained("microsoft/wavlm-large").to(device)
    cmodel.eval()

    # smodel = spk_encoder.load_model(Path(speaker_path), device)
    # smodel = spk_encoder.load_model(Path(f"speaker_encoder/ckpt/pretrained_bak_5805000.pt"), 'cuda')
    smodel = SpeakerEncoder(f"speaker_encoder/ckpt/pretrained_bak_5805000.pt", device)

    return freevc, cmodel, smodel, hps


@torch.no_grad()
def convert(freevc, content, speaker):
    audio = freevc.infer(content, g=speaker)
    audio = audio[0][0].data.cpu().float().numpy()
    return audio, 24000


if __name__ == '__main__':
    freevc_24, cmodel, smodel, hps = get_freevc_models()

    tgt = 'p226_002.wav'
    # src = 'p226_002.wav'
    src = 'p225_001.wav'
    device = 'cuda'

    # tgt
    wav_tgt, _ = librosa.load(tgt, sr=hps.data.sampling_rate)
    wav_tgt, _ = librosa.effects.trim(wav_tgt, top_db=20)
    g_tgt = smodel.embed_utterance(wav_tgt)
    g_tgt = torch.from_numpy(g_tgt).unsqueeze(0).to(device)
    # g_tgt = spk_encoder.embed_utterance_batch(torch.tensor(wav_tgt).unsqueeze(0).cuda())

    # src
    wav_src, _ = librosa.load(src, sr=hps.data.sampling_rate)
    wav_src = torch.from_numpy(wav_src).unsqueeze(0).to(device)
    content = cmodel(wav_src).last_hidden_state.transpose(1, 2).to(device)

    output, sr = convert(freevc_24, content, g_tgt)

    sf.write('output.wav', output, sr)