tinyCLAP: Distilling Constrastive Language-Audio Pretrained Models
- URL: http://arxiv.org/abs/2311.14517v3
- Date: Tue, 24 Sep 2024 11:22:04 GMT
- Title: tinyCLAP: Distilling Constrastive Language-Audio Pretrained Models
- Authors: Francesco Paissan, Elisabetta Farella,
- Abstract summary: This paper investigates how we can reduce the complexity of contrastive language-audio pre-trained models.
We derive an unimodal distillation loss from first principles and explore how the dimensionality of the shared, multimodal latent space can be reduced.
TinyCLAP uses only 6% of the original Microsoft CLAP parameters with a minimal reduction (less than 5%) in zero-shot classification performance.
- Score: 2.9619090219410515
- License: http://arxiv.org/licenses/nonexclusive-distrib/1.0/
- Abstract: Contrastive Language-Audio Pretraining (CLAP) became of crucial importance in the field of audio and speech processing. Its employment ranges from sound event detection to text-to-audio generation. However, one of the main limitations is the considerable amount of data required in the training process and the overall computational complexity during inference. This paper investigates how we can reduce the complexity of contrastive language-audio pre-trained models, yielding an efficient model that we call tinyCLAP. We derive an unimodal distillation loss from first principles and explore how the dimensionality of the shared, multimodal latent space can be reduced via pruning. TinyCLAP uses only 6% of the original Microsoft CLAP parameters with a minimal reduction (less than 5%) in zero-shot classification performance across the three sound event detection datasets on which it was tested
Related papers
- CLAIR-A: Leveraging Large Language Models to Judge Audio Captions [73.51087998971418]
evaluating machine-generated audio captions is a complex task that requires considering diverse factors.
We propose CLAIR-A, a simple and flexible method that leverages the zero-shot capabilities of large language models.
In our evaluations, CLAIR-A better predicts human judgements of quality compared to traditional metrics.
arXiv Detail & Related papers (2024-09-19T17:59:52Z) - T-CLAP: Temporal-Enhanced Contrastive Language-Audio Pretraining [38.604112878493396]
Contrastive language-audio pretraining(CLAP) has been developed to align the representations of audio and language.
We introduce T-CLAP, a temporal-enhanced CLAP model, to capture temporal information within audio and text features.
T-CLAP shows improved capability in capturing the temporal relationship of sound events and outperforms state-of-the-art models by a significant margin.
arXiv Detail & Related papers (2024-04-27T07:05:48Z) - Weakly-supervised Automated Audio Captioning via text only training [1.504795651143257]
We propose a weakly-supervised approach to train an AAC model assuming only text data and a pre-trained CLAP model.
We evaluate our proposed method on Clotho and AudioCaps datasets demonstrating its ability to achieve a relative performance of up to $83%$ compared to fully supervised approaches.
arXiv Detail & Related papers (2023-09-21T16:40:46Z) - SLICER: Learning universal audio representations using low-resource
self-supervised pre-training [53.06337011259031]
We present a new Self-Supervised Learning approach to pre-train encoders on unlabeled audio data.
Our primary aim is to learn audio representations that can generalize across a large variety of speech and non-speech tasks.
arXiv Detail & Related papers (2022-11-02T23:45:33Z) - Simple Pooling Front-ends For Efficient Audio Classification [56.59107110017436]
We show that eliminating the temporal redundancy in the input audio features could be an effective approach for efficient audio classification.
We propose a family of simple pooling front-ends (SimPFs) which use simple non-parametric pooling operations to reduce the redundant information.
SimPFs can achieve a reduction in more than half of the number of floating point operations for off-the-shelf audio neural networks.
arXiv Detail & Related papers (2022-10-03T14:00:41Z) - Self-Supervised Learning for speech recognition with Intermediate layer
supervision [52.93758711230248]
We propose Intermediate Layer Supervision for Self-Supervised Learning (ILS-SSL)
ILS-SSL forces the model to concentrate on content information as much as possible by adding an additional SSL loss on the intermediate layers.
Experiments on LibriSpeech test-other set show that our method outperforms HuBERT significantly.
arXiv Detail & Related papers (2021-12-16T10:45:05Z) - Peer Collaborative Learning for Polyphonic Sound Event Detection [3.325054486984015]
This paper describes that semi-supervised learning called peer collaborative learning (PCL) can be applied to the polyphonic sound event detection task.
We evaluated the proposed PCL model using the DCASE 2019 Task 4 datasets and achieved an F1-score improvement of about 10% compared to the baseline model.
arXiv Detail & Related papers (2021-10-07T14:47:11Z) - Test-Time Adaptation Toward Personalized Speech Enhancement: Zero-Shot
Learning with Knowledge Distillation [26.39206098000297]
We propose a novel personalized speech enhancement method to adapt a compact denoising model to the test-time specificity.
Our goal in this test-time adaptation is to utilize no clean speech target of the test speaker.
Instead of the missing clean utterance target, we distill the more advanced denoising results from an overly large teacher model.
arXiv Detail & Related papers (2021-05-08T00:42:03Z) - Target-Speaker Voice Activity Detection: a Novel Approach for
Multi-Speaker Diarization in a Dinner Party Scenario [51.50631198081903]
We propose a novel Target-Speaker Voice Activity Detection (TS-VAD) approach.
TS-VAD directly predicts an activity of each speaker on each time frame.
Experiments on the CHiME-6 unsegmented data show that TS-VAD achieves state-of-the-art results.
arXiv Detail & Related papers (2020-05-14T21:24:56Z) - You Do Not Need More Data: Improving End-To-End Speech Recognition by
Text-To-Speech Data Augmentation [59.31769998728787]
We build our TTS system on an ASR training database and then extend the data with synthesized speech to train a recognition model.
Our system establishes a competitive result for end-to-end ASR trained on LibriSpeech train-clean-100 set with WER 4.3% for test-clean and 13.5% for test-other.
arXiv Detail & Related papers (2020-05-14T17:24:57Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.