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# Copyright (c) 2023 Amphion. | |
# | |
# This source code is licensed under the MIT license found in the | |
# LICENSE file in the root directory of this source tree. | |
import argparse | |
import os | |
import torch | |
import soundfile as sf | |
import numpy as np | |
from models.tts.naturalspeech2.ns2 import NaturalSpeech2 | |
from encodec import EncodecModel | |
from encodec.utils import convert_audio | |
from utils.util import load_config | |
from text import text_to_sequence | |
from text.cmudict import valid_symbols | |
from text.g2p import preprocess_english, read_lexicon | |
import torchaudio | |
class NS2Inference: | |
def __init__(self, args, cfg): | |
self.cfg = cfg | |
self.args = args | |
self.model = self.build_model() | |
self.codec = self.build_codec() | |
self.symbols = valid_symbols + ["sp", "spn", "sil"] + ["<s>", "</s>"] | |
self.phone2id = {s: i for i, s in enumerate(self.symbols)} | |
self.id2phone = {i: s for s, i in self.phone2id.items()} | |
def build_model(self): | |
model = NaturalSpeech2(self.cfg.model) | |
model.load_state_dict( | |
torch.load( | |
os.path.join(self.args.checkpoint_path, "pytorch_model.bin"), | |
map_location="cpu", | |
) | |
) | |
model = model.to(self.args.device) | |
return model | |
def build_codec(self): | |
encodec_model = EncodecModel.encodec_model_24khz() | |
encodec_model = encodec_model.to(device=self.args.device) | |
encodec_model.set_target_bandwidth(12.0) | |
return encodec_model | |
def get_ref_code(self): | |
ref_wav_path = self.args.ref_audio | |
ref_wav, sr = torchaudio.load(ref_wav_path) | |
ref_wav = convert_audio( | |
ref_wav, sr, self.codec.sample_rate, self.codec.channels | |
) | |
ref_wav = ref_wav.unsqueeze(0).to(device=self.args.device) | |
with torch.no_grad(): | |
encoded_frames = self.codec.encode(ref_wav) | |
ref_code = torch.cat([encoded[0] for encoded in encoded_frames], dim=-1) | |
# print(ref_code.shape) | |
ref_mask = torch.ones(ref_code.shape[0], ref_code.shape[-1]).to(ref_code.device) | |
# print(ref_mask.shape) | |
return ref_code, ref_mask | |
def inference(self): | |
ref_code, ref_mask = self.get_ref_code() | |
lexicon = read_lexicon(self.cfg.preprocess.lexicon_path) | |
phone_seq = preprocess_english(self.args.text, lexicon) | |
print(phone_seq) | |
phone_id = np.array( | |
[ | |
*map( | |
self.phone2id.get, | |
phone_seq.replace("{", "").replace("}", "").split(), | |
) | |
] | |
) | |
phone_id = torch.from_numpy(phone_id).unsqueeze(0).to(device=self.args.device) | |
print(phone_id) | |
x0, prior_out = self.model.inference( | |
ref_code, phone_id, ref_mask, self.args.inference_step | |
) | |
print(prior_out["dur_pred"]) | |
print(prior_out["dur_pred_round"]) | |
print(torch.sum(prior_out["dur_pred_round"])) | |
latent_ref = self.codec.quantizer.vq.decode(ref_code.transpose(0, 1)) | |
rec_wav = self.codec.decoder(x0) | |
# ref_wav = self.codec.decoder(latent_ref) | |
os.makedirs(self.args.output_dir, exist_ok=True) | |
sf.write( | |
"{}/{}.wav".format( | |
self.args.output_dir, self.args.text.replace(" ", "_", 100) | |
), | |
rec_wav[0, 0].detach().cpu().numpy(), | |
samplerate=24000, | |
) | |
def add_arguments(parser: argparse.ArgumentParser): | |
parser.add_argument( | |
"--ref_audio", | |
type=str, | |
default="", | |
help="Reference audio path", | |
) | |
parser.add_argument( | |
"--device", | |
type=str, | |
default="cuda", | |
) | |
parser.add_argument( | |
"--inference_step", | |
type=int, | |
default=200, | |
help="Total inference steps for the diffusion model", | |
) | |