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import torch
import torch.nn as nn
from transformers import Wav2Vec2Processor
from transformers.models.wav2vec2.modeling_wav2vec2 import (
Wav2Vec2Model,
Wav2Vec2PreTrainedModel,
)
import os
import librosa
import numpy as np
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
# load model from hub
device = 'cuda' if torch.cuda.is_available() else "cpu"
model_name = 'audeering/wav2vec2-large-robust-12-ft-emotion-msp-dim'
processor = Wav2Vec2Processor.from_pretrained(model_name)
model = EmotionModel.from_pretrained(model_name).to(device)
def process_func(
x: np.ndarray,
sampling_rate: int,
embeddings: bool = False,
) -> np.ndarray:
r"""Predict emotions or extract embeddings from raw audio signal."""
# run through processor to normalize signal
# always returns a batch, so we just get the first entry
# then we put it on the device
y = processor(x, sampling_rate=sampling_rate)
y = y['input_values'][0]
y = torch.from_numpy(y).to(device)
y = y.unsqueeze(0)
# 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 disp(rootpath, wavname):
# wav, sr = librosa.load(f"{rootpath}/{wavname}", 16000)
# display(ipd.Audio(wav, rate=sr))
rootpath = "dataset"
embs = []
wavnames = []
def extract_dir(path):
rootpath = path
for idx, wavname in enumerate(os.listdir(rootpath)):
wav, sr =librosa.load(f"{rootpath}/{wavname}", sr=16000)
emb = process_func(np.expand_dims(wav, 0), sr, embeddings=True)
embs.append(emb)
wavnames.append(wavname)
np.save(f"{rootpath}/{wavname}.emo.npy", emb.squeeze(0))
print(idx, wavname)
def extract_wav(path):
wav, sr = librosa.load(path, sr=16000)
emb = process_func(np.expand_dims(wav, 0), sr, embeddings=True)
return emb
if __name__ == '__main__':
# for spk in ["serena", "koni", "nyaru","shanoa", "mana"]:
for spk in ["dubbingx"]:
extract_dir(f"dataset/{spk}")
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