Spaces:
Sleeping
Sleeping
import torch | |
import torch.nn as nn | |
from torch.utils.data import Dataset | |
from torch.utils.data import DataLoader | |
from transformers import Wav2Vec2Processor | |
from transformers.models.wav2vec2.modeling_wav2vec2 import ( | |
Wav2Vec2Model, | |
Wav2Vec2PreTrainedModel, | |
) | |
import librosa | |
import numpy as np | |
import argparse | |
from config import config | |
import utils | |
import os | |
from tqdm import tqdm | |
class RegressionHead(nn.Module): | |
r"""Classification head.""" | |
def __init__(self, config): | |
super().__init__() | |
self.dense = nn.Linear(config.hidden_size, config.hidden_size) | |
self.dropout = nn.Dropout(config.final_dropout) | |
self.out_proj = nn.Linear(config.hidden_size, config.num_labels) | |
def forward(self, features, **kwargs): | |
x = features | |
x = self.dropout(x) | |
x = self.dense(x) | |
x = torch.tanh(x) | |
x = self.dropout(x) | |
x = self.out_proj(x) | |
return x | |
class EmotionModel(Wav2Vec2PreTrainedModel): | |
r"""Speech emotion classifier.""" | |
def __init__(self, config): | |
super().__init__(config) | |
self.config = config | |
self.wav2vec2 = Wav2Vec2Model(config) | |
self.classifier = RegressionHead(config) | |
self.init_weights() | |
def forward( | |
self, | |
input_values, | |
): | |
outputs = self.wav2vec2(input_values) | |
hidden_states = outputs[0] | |
hidden_states = torch.mean(hidden_states, dim=1) | |
logits = self.classifier(hidden_states) | |
return hidden_states, logits | |
class AudioDataset(Dataset): | |
def __init__(self, list_of_wav_files, sr, processor): | |
self.list_of_wav_files = list_of_wav_files | |
self.processor = processor | |
self.sr = sr | |
def __len__(self): | |
return len(self.list_of_wav_files) | |
def __getitem__(self, idx): | |
wav_file = self.list_of_wav_files[idx] | |
audio_data, _ = librosa.load(wav_file, sr=self.sr) | |
processed_data = self.processor(audio_data, sampling_rate=self.sr)[ | |
"input_values" | |
][0] | |
return torch.from_numpy(processed_data) | |
model_name = "./emotional/wav2vec2-large-robust-12-ft-emotion-msp-dim" | |
processor = Wav2Vec2Processor.from_pretrained(model_name) | |
model = EmotionModel.from_pretrained(model_name) | |
def process_func( | |
x: np.ndarray, | |
sampling_rate: int, | |
model: EmotionModel, | |
processor: Wav2Vec2Processor, | |
device: str, | |
embeddings: bool = False, | |
) -> np.ndarray: | |
r"""Predict emotions or extract embeddings from raw audio signal.""" | |
model = model.to(device) | |
y = processor(x, sampling_rate=sampling_rate) | |
y = y["input_values"][0] | |
y = torch.from_numpy(y).unsqueeze(0).to(device) | |
# run through model | |
with torch.no_grad(): | |
y = model(y)[0 if embeddings else 1] | |
# convert to numpy | |
y = y.detach().cpu().numpy() | |
return y | |
def get_emo(path): | |
wav, sr = librosa.load(path, 16000) | |
device = config.bert_gen_config.device | |
return process_func( | |
np.expand_dims(wav, 0).astype(np.float), | |
sr, | |
model, | |
processor, | |
device, | |
embeddings=True, | |
).squeeze(0) | |
if __name__ == "__main__": | |
parser = argparse.ArgumentParser() | |
parser.add_argument( | |
"-c", "--config", type=str, default=config.bert_gen_config.config_path | |
) | |
parser.add_argument( | |
"--num_processes", type=int, default=config.bert_gen_config.num_processes | |
) | |
args, _ = parser.parse_known_args() | |
config_path = args.config | |
hps = utils.get_hparams_from_file(config_path) | |
device = config.bert_gen_config.device | |
model_name = "./emotional/wav2vec2-large-robust-12-ft-emotion-msp-dim" | |
processor = ( | |
Wav2Vec2Processor.from_pretrained(model_name) | |
if processor is None | |
else processor | |
) | |
model = ( | |
EmotionModel.from_pretrained(model_name).to(device) | |
if model is None | |
else model.to(device) | |
) | |
lines = [] | |
with open(hps.data.training_files, encoding="utf-8") as f: | |
lines.extend(f.readlines()) | |
with open(hps.data.validation_files, encoding="utf-8") as f: | |
lines.extend(f.readlines()) | |
wavnames = [line.split("|")[0] for line in lines] | |
dataset = AudioDataset(wavnames, 16000, processor) | |
data_loader = DataLoader(dataset, batch_size=1, shuffle=False, num_workers=16) | |
with torch.no_grad(): | |
for i, data in tqdm(enumerate(data_loader), total=len(data_loader)): | |
wavname = wavnames[i] | |
emo_path = wavname.replace(".wav", ".emo.npy") | |
if os.path.exists(emo_path): | |
continue | |
emb = model(data.to(device))[0].detach().cpu().numpy() | |
np.save(emo_path, emb) | |
print("Emo vec 生成完毕!") | |