Audio Classification
Transformers
PyTorch
hubert
feature-extraction
music
audio
speech
audio-representation-learning
arch-benchmark
general-audio
Instructions to use ALM/hubert-large-audioset with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use ALM/hubert-large-audioset with Transformers:
# Use a pipeline as a high-level helper from transformers import pipeline pipe = pipeline("audio-classification", model="ALM/hubert-large-audioset")# Load model directly from transformers import AutoProcessor, AutoModel processor = AutoProcessor.from_pretrained("ALM/hubert-large-audioset") model = AutoModel.from_pretrained("ALM/hubert-large-audioset", device_map="auto") - Notebooks
- Google Colab
- Kaggle
| license: cc-by-nc-sa-4.0 | |
| pipeline_tag: audio-classification | |
| tags: | |
| - music | |
| - audio | |
| - speech | |
| - audio-representation-learning | |
| - arch-benchmark | |
| - general-audio | |
| # Model Card: Pre-trained Audio Representation Models on AudioSet | |
| ## Overview | |
| This model card presents information about pre-trained audio representation models released by ALM. These models are pre-trained on the full AudioSet dataset and are intended for general-purpose Audio Representation Learning (ARL) tasks. | |
| ## Models | |
| ### 1. [ALM/hubert-base-audioset](https://huggingface.co/ALM/hubert-base-audioset) | |
| - **Architecture**: HuBERT (Hubert-Base) transformer-based model | |
| - **Description**: This model is based on the HuBERT architecture, pre-trained on the full AudioSet dataset. | |
| ### 2. [ALM/hubert-large-audioset](https://huggingface.co/ALM/hubert-large-audioset) | |
| - **Architecture**: HuBERT (Hubert-Large) transformer-based model | |
| - **Description**: Similar to the hubert-base-audioset model, this variant is larger in size, providing increased capacity for capturing audio representations from the full AudioSet dataset. | |
| ### 3. [ALM/wav2vec2-base-audioset](https://huggingface.co/ALM/wav2vec2-base-audioset) | |
| - **Architecture**: Wav2Vec 2.0 (Wav2Vec2-Base) transformer-based model | |
| - **Description**: This model is based on the Wav2Vec 2.0 architecture, trained on the full AudioSet dataset using SSL with CPC. It offers a different approach to audio representation learning compared to the HuBERT models. | |
| ### 4. [ALM/wav2vec2-large-audioset](https://huggingface.co/ALM/wav2vec2-large-audioset) | |
| - **Architecture**: Wav2Vec 2.0 (Wav2Vec2-Large) transformer-based model | |
| - **Description**: Similar to the wav2vec2-base-audioset model, this variant is larger in size, providing enhanced capacity for learning audio representations from the full AudioSet dataset. | |
| ## Intended Use | |
| These pre-trained models are intended for a wide range of ARL tasks, including but not limited to speech recognition, music classification, and acoustic event detection. They serve as powerful tools for feature extraction and can be fine-tuned on task-specific datasets for downstream applications. | |
| It's important to note that while these models offer versatility across various audio domains, their performance in speech-related tasks may be relatively lower compared to specialized models such as the original Wav2Vec and HuBERT models. | |
| This is due to the diverse nature of the AudioSet dataset used for pre-training, which includes a wide range of audio sources beyond speech. | |
| ## Limitations and Considerations | |
| - The models are pre-trained on the full AudioSet dataset, which may not cover all possible audio domains comprehensively. | |
| - Fine-tuning on domain-specific data may be necessary to achieve optimal performance for certain tasks. | |
| - Computational resources may be required for deploying and fine-tuning these models, especially the larger variants. | |
| ## Citation | |
| If you use these pre-trained models in your work, please cite the following | |
| ```bib | |
| @INPROCEEDINGS{ARCH, | |
| author={La Quatra, Moreno and Koudounas, Alkis and Vaiani, Lorenzo and Baralis, Elena and Cagliero, Luca and Garza, Paolo and Siniscalchi, Sabato Marco}, | |
| booktitle={2024 IEEE International Conference on Acoustics, Speech, and Signal Processing Workshops (ICASSPW)}, | |
| title={Benchmarking Representations for Speech, Music, and Acoustic Events}, | |
| year={2024}, | |
| pages={505-509}, | |
| keywords={Representation learning; Systematics; Conferences; Benchmark testing; Signal processing; Acoustics; Data models; Audio Representation Learning; Benchmark; Pre-trained Models; Self-Supervised Learning}, | |
| doi={10.1109/ICASSPW62465.2024.10625960} | |
| } | |
| ``` | |
| [arXiv version: arxiv.org/abs/2405.00934](arxiv.org/abs/2405.00934) |