Efficient Integration of Self-Supervised Speech Representations into Automatic Speech Recognition Systems

核心概念

Two efficient approaches to incorporate representations from pre-trained self-supervised speech models into end-to-end ASR architectures, resulting in faster training and significant performance gains.

要約

The paper proposes two approaches to efficiently integrate representations from pre-trained self-supervised speech models (such as Wav2Vec and HuBERT) into end-to-end automatic speech recognition (ASR) architectures:

1. Subsampled Framewise Addition (SFA): This simple, parameterless approach performs elementwise addition of the subsampled self-supervised representations with the fbank features. While efficient, it has limitations in handling cases where the lengths of the two sequences differ significantly.

2. Cross Attention (CA): To address the limitations of SFA, the authors introduce a cross-attention layer that allows the model to dynamically learn the alignment between the fbank and self-supervised representations.

The authors evaluate these approaches on the Librispeech and Tedlium datasets, demonstrating significant reductions in word error rate (WER) compared to baseline conformer and transformer models. Key findings include:

• HuBERT representations consistently outperform Wav2Vec representations.
• The cross-attention approach outperforms the subsampled framewise addition method.
• The proposed architectures achieve faster convergence, requiring only 10 epochs to outperform the baseline.
• Even with a significant reduction in the number of encoder layers (up to 80%), the models still perform much better than the baseline.
• Visualization of the attention scores shows that the model has learned meaningful alignments between the fbank and self-supervised representations.
• The approach is effective on unseen datasets, with the HuBERT-LARGE model performing the best on the Tedlium dataset.

Overall, the paper presents efficient and effective methods to incorporate self-supervised speech representations into end-to-end ASR systems, leading to faster training and improved performance.

統計

The paper reports the following key metrics:

Word error rate (WER) on the Librispeech-100 and Tedlium2 datasets
Number of model parameters
Training time

引用

"While directly incorporating HuBERT or Wav2vec as a frontend/Encoder is bound to perform better, our approach results in a model that is both faster in training and smaller in size at the cost of a slight degradation in performance."
"Even after reducing the number of encoder layers by 80%, we find that the model still performs much better despite having only half as many parameters and training time as the baseline model."

抽出されたキーインサイト

Efficient infusion of self-supervised representations in Automatic Speech Recognition

by Darshan Prab... 場所 arxiv.org 04-22-2024

https://arxiv.org/pdf/2404.12628.pdf

深掘り質問

How can the proposed approaches be extended to incorporate representations from multiple self-supervised models in a more seamless and effective manner

The proposed approaches for incorporating representations from multiple self-supervised models can be extended by implementing a more sophisticated fusion mechanism that can effectively combine the features from different models. One way to achieve this is by introducing a hierarchical fusion strategy where the representations from each model are first processed independently and then fused at different levels of abstraction. For example, the lower-level features can be combined early in the architecture, while higher-level features can be integrated closer to the output layer. This hierarchical fusion can help capture complementary information from each model and enhance the overall representation quality.
Additionally, employing a dynamic fusion mechanism that adaptively weights the contributions of each model based on the input data can further improve the integration process. By introducing learnable parameters that control the fusion process, the model can effectively leverage the strengths of each self-supervised model based on the context of the input audio. This dynamic fusion approach can enhance the flexibility and adaptability of the architecture when incorporating multiple representations.
Furthermore, exploring ensemble techniques where the predictions from individual models are combined at the output stage can also be beneficial. By training multiple models with different self-supervised representations and aggregating their outputs using techniques like averaging or stacking, the overall performance can be boosted. This ensemble approach can help mitigate the limitations of individual models and improve the robustness of the ASR system.

What are the potential limitations or drawbacks of relying on pre-trained self-supervised representations, and how can they be addressed

While leveraging pre-trained self-supervised representations offers significant advantages in terms of performance and efficiency, there are potential limitations and drawbacks that need to be addressed. One key limitation is the domain specificity of the pre-trained models, as they may not capture all the nuances and characteristics of the target ASR task. This can lead to suboptimal performance, especially in out-of-domain or low-resource scenarios. To mitigate this limitation, fine-tuning the self-supervised models on domain-specific data or incorporating domain adaptation techniques can help align the representations with the target task more effectively.
Another drawback is the computational overhead associated with using large pre-trained models, which can hinder deployment on low-resource or edge devices. To address this, techniques like model distillation, quantization, and pruning can be applied to reduce the model size and computational complexity while preserving performance. By optimizing the architecture for efficiency without compromising accuracy, the ASR system can be made more suitable for resource-constrained environments.
Moreover, the lack of interpretability in self-supervised representations can pose challenges in understanding the underlying features learned by the models. To overcome this limitation, techniques such as attention visualization, feature attribution methods, and interpretability frameworks can be employed to gain insights into the representation learning process. By enhancing the interpretability of the models, researchers and practitioners can better analyze and optimize the performance of the ASR system.

Given the focus on efficiency and performance, how might these techniques be applied to low-resource or edge-device ASR scenarios

To apply the proposed techniques to low-resource or edge-device ASR scenarios, a few key considerations need to be taken into account. Firstly, optimizing the architecture for efficiency by reducing the model size, computational complexity, and memory footprint is crucial. This can be achieved through techniques like model compression, quantization, and knowledge distillation, which enable the ASR system to run efficiently on devices with limited resources.
Additionally, leveraging hardware accelerators such as GPUs, TPUs, or specialized AI chips can enhance the inference speed and performance of the ASR system on edge devices. By utilizing hardware optimizations and parallel processing capabilities, the computational burden can be alleviated, enabling real-time speech recognition even on low-power devices.
Furthermore, exploring on-device training and adaptation techniques can be beneficial for low-resource scenarios where labeled data is scarce. By incorporating online learning strategies, transfer learning, and active learning approaches, the ASR system can adapt to new environments, accents, or languages with minimal supervision. This adaptive learning capability can improve the robustness and accuracy of the system in diverse settings, making it more suitable for real-world applications in low-resource contexts.