ExHuBERT

ExHuBERT: Enhancing HuBERT Through Block Extension and Fine-Tuning on 37 Emotion Datasets

Authors: Shahin Amiriparian, Filip Packań, Maurice Gerczuk, Björn W. Schuller

Fine-tuned and backbone extended HuBERT Large on EmoSet++, comprising 37 datasets, totaling 150,907 samples and spanning a cumulative duration of 119.5 hours. The model is expecting a 3 second long raw waveform resampled to 16 kHz. The original 6 Ouput classes are combinations of low/high arousal and negative/neutral/positive valence. Further details are available in the corresponding paper.

EmoSet++ subsets used for fine-tuning the model:

Usage

import torch
import torch.nn as nn
from transformers import AutoModelForAudioClassification, Wav2Vec2FeatureExtractor


# CONFIG and MODEL SETUP
model_name = 'amiriparian/ExHuBERT'
feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("facebook/hubert-base-ls960")
model = AutoModelForAudioClassification.from_pretrained(model_name, trust_remote_code=True,
                                                        revision="b158d45ed8578432468f3ab8d46cbe5974380812")

# Freezing half of the encoder for further transfer learning
model.freeze_og_encoder()

sampling_rate = 16000
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
model = model.to(device)



# Example application from a local audiofile
import numpy as np
import librosa
import torch.nn.functional as F
# Sample taken from the Toronto emotional speech set (TESS) https://tspace.library.utoronto.ca/handle/1807/24487
waveform, sr_wav = librosa.load("YAF_date_angry.wav")
# Max Padding to 3 Seconds at 16k sampling rate for the best results
waveform = feature_extractor(waveform, sampling_rate=sampling_rate,padding = 'max_length',max_length = 48000)
waveform = waveform['input_values'][0]
waveform = waveform.reshape(1, -1)
waveform = torch.from_numpy(waveform).to(device)
with torch.no_grad():
    output = model(waveform)
    output = F.softmax(output.logits, dim = 1)
    output = output.detach().cpu().numpy().round(2)
    print(output)

    # [[0.      0.      0.      1.      0.      0.]]
    #          Low          |          High                 Arousal
    # Neg.     Neut.   Pos. |  Neg.    Neut.   Pos          Valence
    # Disgust, Neutral, Kind| Anger, Surprise, Joy          Example emotions

Example of How to Train the Model for Transfer Learning

The datasets used for showcasing are EmoDB and IEMOCAP from the HuggingFace Hub. As noted above, the model has seen both datasets before.

import pandas as pd
import torch
import torch.nn as nn
from torch.utils.data import Dataset, DataLoader
import librosa
import io
from transformers import AutoModelForAudioClassification, Wav2Vec2FeatureExtractor

# CONFIG and MODEL SETUP
model_name = 'amiriparian/ExHuBERT'
feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("facebook/hubert-base-ls960")
model = AutoModelForAudioClassification.from_pretrained(model_name, trust_remote_code=True,
                                                        revision="b158d45ed8578432468f3ab8d46cbe5974380812")

# Replacing Classifier layer
model.classifier = nn.Linear(in_features=256, out_features=7)
# Freezing the original encoder layers and feature encoder (as in the paper) for further transfer learning
model.freeze_og_encoder()
model.freeze_feature_encoder()
model.train()

device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
model = model.to(device)

# Define a custom dataset class
class EmotionDataset(Dataset):
    def __init__(self, dataframe, feature_extractor, max_length):
        self.dataframe = dataframe
        self.feature_extractor = feature_extractor
        self.max_length = max_length

    def __len__(self):
        return len(self.dataframe)

    def __getitem__(self, idx):
        row = self.dataframe.iloc[idx]
        # emotion = torch.tensor(row['label'], dtype=torch.int64)  # For the IEMOCAP example
        emotion = torch.tensor(row['emotion'], dtype=torch.int64)  # EmoDB specific

        # Decode audio bytes from the Huggingface dataset with librosa
        audio_bytes = row['audio']['bytes']
        audio_buffer = io.BytesIO(audio_bytes)
        audio_data, samplerate = librosa.load(audio_buffer, sr=16000)

        # Use the feature extractor to preprocess the audio. Padding/Truncating to 3 seconds gives better results
        audio_features = self.feature_extractor(audio_data, sampling_rate=16000, return_tensors="pt", padding="max_length",
                                                truncation=True, max_length=self.max_length)

        audio = audio_features['input_values'].squeeze(0)
        return audio, emotion

# Load your DataFrame. Samples are shown for EmoDB and IEMOCAP from the Huggingface Hub
df = pd.read_parquet("hf://datasets/renumics/emodb/data/train-00000-of-00001-cf0d4b1ae18136ff.parquet")
# splits = {'session1': 'data/session1-00000-of-00001-04e11ca668d90573.parquet', 'session2': 'data/session2-00000-of-00001-f6132100b374cb18.parquet', 'session3': 'data/session3-00000-of-00001-6e102fcb5c1126b4.parquet', 'session4': 'data/session4-00000-of-00001-e39531a7c694b50d.parquet', 'session5': 'data/session5-00000-of-00001-03769060403172ce.parquet'}
# df = pd.read_parquet("hf://datasets/Zahra99/IEMOCAP_Audio/" + splits["session1"])

# Dataset and DataLoader
dataset = EmotionDataset(df, feature_extractor, max_length=3 * 16000)
dataloader = DataLoader(dataset, batch_size=4, shuffle=True)

# Training setup
criterion = nn.CrossEntropyLoss()
lr = 1e-5
non_frozen_parameters = [p for p in model.parameters() if p.requires_grad]
optim = torch.optim.AdamW(non_frozen_parameters, lr=lr, betas=(0.9, 0.999), eps=1e-08)

# Function to calculate accuracy
def calculate_accuracy(outputs, targets):
    _, predicted = torch.max(outputs, 1)
    correct = (predicted == targets).sum().item()
    return correct / targets.size(0)

# Training loop
num_epochs = 3
for epoch in range(num_epochs):
    model.train()
    total_loss = 0.0
    total_correct = 0
    total_samples = 0
    for batch_idx, (inputs, targets) in enumerate(dataloader):
        inputs, targets = inputs.to(device), targets.to(device)

        optim.zero_grad()
        outputs = model(inputs).logits
        loss = criterion(outputs, targets)
        loss.backward()
        optim.step()

        total_loss += loss.item()
        total_correct += (outputs.argmax(1) == targets).sum().item()
        total_samples += targets.size(0)

    epoch_loss = total_loss / len(dataloader)
    epoch_accuracy = total_correct / total_samples
    print(f'Epoch [{epoch + 1}/{num_epochs}], Average Loss: {epoch_loss:.4f}, Average Accuracy: {epoch_accuracy:.4f}')

# Example outputs:
# Epoch [3/3], Average Loss: 0.4572, Average Accuracy: 0.8249 for IEMOCAP
# Epoch [3/3], Average Loss: 0.1511, Average Accuracy: 0.9850 for EmoDB

Citation Info

ExHuBERT has been accepted for presentation at INTERSPEECH 2024.

@inproceedings{amiriparian24_interspeech,
  title     = {ExHuBERT: Enhancing HuBERT Through Block Extension and Fine-Tuning on 37 Emotion Datasets},
  author    = {Shahin Amiriparian and Filip Packań and Maurice Gerczuk and Björn W. Schuller},
  year      = {2024},
  booktitle = {Interspeech 2024},
  pages     = {2635--2639},
  doi       = {10.21437/Interspeech.2024-280},
  issn      = {2958-1796},
}

<!-- ``` @inproceedings{amiriparian24_interspeech, title = {ExHuBERT: Enhancing HuBERT Through Block Extension and Fine-Tuning on 37 Emotion Datasets}, author = {Shahin Amiriparian and Filip Packań and Maurice Gerczuk and Björn W. Schuller}, year = {2024}, booktitle = {Interspeech 2024}, pages = {2635--2639}, doi = {10.21437/Interspeech.2024-280}, } ``` -->

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