Self-supervised learning (SSL) is a powerful technique for learning representations from unlabeled data. Transformer based models such as HuBERT, which consist a feature extractor and transformer layers, are leading the field in the speech domain. SSL models are fine-tuned on a wide range of downstream tasks, which involves re-training the majority of the model for each task. Previous studies have introduced applying adapters, which are small lightweight modules commonly used in Natural Language Processing (NLP) to adapt pre-trained models to new tasks. However, such efficient tuning techniques only provide adaptation at the transformer layer, but failed to perform adaptation at the feature extractor. In this paper, we propose CHAPTER, an efficient tuning method specifically designed for SSL speech model, by applying CNN adapters at the feature extractor. Using this method, we can only fine-tune fewer than 5% of parameters per task compared to fully fine-tuning and achieve better and more stable performance. We empirically found that adding CNN adapters to the feature extractor can help the adaptation on emotion and speaker tasks. For instance, the accuracy of SID is improved from 87.71 to 91.56, and the accuracy of ER is improved by 5%.

Self-supervised pre-training of a speech foundation model, followed by supervised fine-tuning, has shown impressive quality improvements on automatic speech recognition (ASR) tasks. Fine-tuning separate foundation models for many downstream tasks are expensive since the foundation model is usually very big. Parameter-efficient fine-tuning methods (e.g. adapter, sparse update methods) offer an alternative paradigm where a small set of parameters are updated to adapt the foundation model to new tasks. However, these methods still suffer from a high computational memory cost and slow training speed because they require backpropagation through the entire neural network at each step. In the paper, we analyze the performance of features at different layers of a foundation model on the speech recognition task and propose a novel hierarchical feature fusion method for resource-efficient transfer learning from speech foundation models. Experimental results show that the proposed method can achieve better performance on speech recognition task than existing algorithms with fewer number of trainable parameters, less computational memory cost and faster training speed. After combining with Adapters at all layers, the proposed method can achieve the same performance as fine-tuning the whole model with $97\%$ fewer trainable encoder parameters and $53\%$ faster training speed.

尽管参数有效调整（PET）方法在自然语言处理（NLP）任务上显示出巨大的潜力，但其有效性仍然对计算机视觉（CV）任务的大规模转向进行了研究。本文提出了Conv-Adapter，这是一种专为CONCNET设计的PET模块。 Conv-Adapter具有轻巧的，可转让的域和架构，不合时宜，并且在不同的任务上具有广义性能。当转移下游任务时，Conv-Adapter将特定于任务的特征调制到主链的中间表示，同时保持预先训练的参数冻结。通过仅引入少量可学习的参数，例如，仅3.5％的RESNET50的完整微调参数，Conv-Adapter优于先前的宠物基线方法，并实现可比性或超过23个分类任务的全面调查的性能。它还在几乎没有分类的情况下表现出卓越的性能，平均利润率为3.39％。除分类外，Conv-Adapter可以推广到检测和细分任务，其参数降低了50％以上，但性能与传统的完整微调相当。

具有数百万参数的基于变压器的预训练模型需要大量存储。最近的方法通过培训适配器解决了这一缺点，但是这些方法仍然需要相对较大的参数。在这项研究中，提出了一种令人惊讶的简单但有效的适配器体系结构的Adapterbias。AdapterBias向变压器层的隐藏输出添加了代币依赖性转移，以适应仅使用向量和线性层的下游任务。进行了广泛的实验，以证明适配性的有效性。实验表明，与先前的作品相比，我们提出的方法可以大大减少可训练的参数，而任务性能与微调的预训练模型相比最小。我们进一步发现，适应性比亚斯自动学习以将更重要的表示形式分配给与任务相关的代币转移。

大规模的语音自我监督学习（SSL）已经出现到语音处理的主要领域，但是，由于其巨大规模而引起的计算成本问题是对学术界的高障碍。此外，语音SSL模型的现有蒸馏技术通过减少层来压缩模型，从而在语言模式识别任务（例如音素识别（PR））中引起性能降解。在本文中，我们提出了Fithubert，它几乎在几乎所有模型组件中都使尺寸较薄，并且与先前的语音SSL蒸馏作品相比，层层更深。此外，我们采用缩短时间来加快推理时间，并提出一种基于提示的蒸馏方法，以减少性能降解。与休伯特相比，我们的方法将模型降低到23.8％，推理时间为35.9％。此外，我们在优越的基准上达到了12.1％的单词错误率和13.3％的音素错误率，这比先前的工作优越。

Fine-tuning large pre-trained models is an effective transfer mechanism in NLP. However, in the presence of many downstream tasks, fine-tuning is parameter inefficient: an entire new model is required for every task. As an alternative, we propose transfer with adapter modules. Adapter modules yield a compact and extensible model; they add only a few trainable parameters per task, and new tasks can be added without revisiting previous ones. The parameters of the original network remain fixed, yielding a high degree of parameter sharing. To demonstrate adapter's effectiveness, we transfer the recently proposed BERT Transformer model to 26 diverse text classification tasks, including the GLUE benchmark. Adapters attain near state-of-the-art performance, whilst adding only a few parameters per task. On GLUE, we attain within 0.4% of the performance of full fine-tuning, adding only 3.6% parameters per task. By contrast, fine-tuning trains 100% of the parameters per task.

学习高级语音表征的自学学习（SSL）一直是在低资源环境中构建自动语音识别（ASR）系统的一种流行方法。但是，文献中提出的共同假设是，可以使用可用于SSL预训练的相同域或语言的大量未标记数据，我们承认，在现实世界中，这是不可行的。在本文中，作为Interspeech Gram Vaani ASR挑战的一部分，我们尝试研究域，语言，数据集大小和上游训练SSL数据对最终性能下游ASR任务的效果。我们还建立在持续的训练范式的基础上，以研究使用SSL训练的模型所拥有的先验知识的效果。广泛的实验和研究表明，ASR系统的性能易受用于SSL预训练的数据。它们的性能随着相似性和预训练数据量的增加而提高。我们认为，我们的工作将有助于语音社区在低资源环境中建立更好的ASR系统，并引导研究改善基于SSL的语音系统预培训的概括。

通过微调将大规模的预训练语言模型适应下游任务是实现NLP基准测试最先进性能的标准方法。然而，微调具有数百万或数十亿个参数的所有重量模型是对低资源设置中不稳定的采样低效，并且浪费，因为它需要为每个任务存储模型的单独副本。最近的工作已经开发了参数高效的微调方法，但这些方法仍然需要相对大量的参数或表现不足标准微调。在这项工作中，我们提出了一种特殊调整大型语言模型的方法，其在任务性能和比率参数之间具有更好的权衡的方法，而不是比上事先工作。 Compacter通过构建适配器，低级优化和参数化超复分乘法层的思想之上来实现这一目标。具体地，Compacter将特定于特定的权重矩阵插入到预估计模型的权重中，这些权重被有效地计算为共享的“慢速”权重和“快速”等级 - 每个Compacter层定义的矩阵之间的矩阵产品的总和。仅通过培训0.047％的预磨料模型的参数，Compacter会在胶水上标准微调和胜过标准微调的标准微调和低资源设置。我们的代码在〜\ url {https://github.com/rabeehk/compacter}上公开使用。

自我监督学习（SSL）在语音识别方面取得了巨大的成功，而有限的探索已尝试完成其他语音处理任务。由于语音信号包含多方面的信息，包括说话者身份，副语言学，口语内容等，学习所有语音任务的通用表示都具有挑战性。为了解决该问题，我们提出了一个新的预培训模型WAVLM，以解决全堆栈的下游语音任务。 Wavlm共同学习了蒙面的语音预测和预训练。通过这种方式，WAVLM不仅可以通过掩盖的语音预测来保持语音内容建模能力，而且还可以通过语音denoing来提高非ASR任务的潜力。此外，WAVLM还采用封闭式的变压器结构的封闭相对位置偏置，以更好地捕获输入语音的序列排序。我们还将培训数据集从60k小时扩展到94K小时。 WAVLM大型在精湛的基准上实现了最先进的性能，并在其代表性基准上为各种语音处理任务带来了重大改进。代码和预培训模型可在https://aka.ms/wavlm上找到。

交叉语言语音适应旨在解决利用多种丰富资源语言来构建低资源目标语言的模型的问题。由于低资源语言具有有限的培训数据，语音识别模型可以容易地过度装备。在本文中，我们建议使用适配器来研究多种适配器的性能，用于参数有效的交叉语音语音适应。基于我们以前的MetaAdapter，隐含地利用适配器，我们提出了一种名为SimAdapter的新算法，用于从Adapters明确学习知识。我们的算法利用了可以轻松集成到变压器结构中的适配器.METAADAPTER利用元学习将一般知识从训练数据转移到测试语言。 SimAdapter旨在使用适配器微调期间了解源语言与目标语言之间的相似性。我们在公共语音数据集中对五种低资源语言进行广泛的实验。结果表明，与强大的全型微调基线相比，我们的MetaAdapter和SimAdapter方法可以将WER减小2.98％和2.55％，只有2.5％和15.5％的培训参数。此外，我们还表明这两种新型算法可以集成，以便更好的性能，相对减少高达3.55％。

Large speech emotion recognition datasets are hard to obtain, and small datasets may contain biases. Deep-net-based classifiers, in turn, are prone to exploit those biases and find shortcuts such as speaker characteristics. These shortcuts usually harm a model's ability to generalize. To address this challenge, we propose a gradient-based adversary learning framework that learns a speech emotion recognition task while normalizing speaker characteristics from the feature representation. We demonstrate the efficacy of our method on both speaker-independent and speaker-dependent settings and obtain new state-of-the-art results on the challenging IEMOCAP dataset.

最近出现了有希望的表现，利用大型预训练的模型来实现各种感兴趣的下游任务。由于模型的规模不断增长，因此，在模型培训和存储方面，基于标准的完整任务适应策略的成本高昂。这导致了参数有效传输学习的新研究方向。但是，现有的尝试通常集中在预训练模型的相同模式（例如图像理解）的下游任务上。这会产生限制，因为在某些特定的方式（例如，视频理解）中，具有足够知识的强大预训练模型较少或不可用。在这项工作中，我们研究了这样一种新型的跨模式转移学习设置，即参数有效的图像到视频传输学习。为了解决此问题，我们为每个视频任务提出了一个新的时空适配器（ST-ADAPTER），以进行参数有效调整。凭借紧凑设计中的内置时空推理能力，ST-ADAPTER可以实现预训练的图像模型，而无需时间知识，以小（〜8％）的每任务参数成本来理解动态视频内容，以大约需要与以前的工作相比，更新参数少20倍。在视频动作识别任务上进行的广泛实验表明，我们的ST-ADAPTER可以匹配甚至优于强大的完整微调策略和最先进的视频模型，同时享受参数效率的优势。

Spoken language understanding (SLU) is a task aiming to extract high-level semantics from spoken utterances. Previous works have investigated the use of speech self-supervised models and textual pre-trained models, which have shown reasonable improvements to various SLU tasks. However, because of the mismatched modalities between speech signals and text tokens, previous methods usually need complex designs of the frameworks. This work proposes a simple yet efficient unsupervised paradigm that connects speech and textual pre-trained models, resulting in an unsupervised speech-to-semantic pre-trained model for various tasks in SLU. To be specific, we propose to use unsupervised automatic speech recognition (ASR) as a connector that bridges different modalities used in speech and textual pre-trained models. Our experiments show that unsupervised ASR itself can improve the representations from speech self-supervised models. More importantly, it is shown as an efficient connector between speech and textual pre-trained models, improving the performances of five different SLU tasks. Notably, on spoken question answering, we reach the state-of-the-art result over the challenging NMSQA benchmark.

Fine-tuning a Pre-trained Language Model (PLM) on a specific downstream task has been a well-known paradigm in Natural Language Processing. However, with the ever-growing size of PLMs, training the entire model on several downstream tasks becomes very expensive and resource-hungry. Recently, different Parameter Efficient Tuning (PET) techniques are proposed to improve the efficiency of fine-tuning PLMs. One popular category of PET methods is the low-rank adaptation methods which insert learnable truncated SVD modules into the original model either sequentially or in parallel. However, low-rank decomposition suffers from limited representation power. In this work, we address this problem using the Kronecker product instead of the low-rank representation. We introduce KronA, a Kronecker product-based adapter module for efficient fine-tuning of Transformer-based PLMs. We apply the proposed methods for fine-tuning T5 on the GLUE benchmark to show that incorporating the Kronecker-based modules can outperform state-of-the-art PET methods.

最近，在大型文本语料库上预先培训的微调语言模型已经为Vision-and Langual（V＆L）任务以及纯语言任务提供了巨大的改进。但是，微调预训练模型的整个参数集变得不切实际，因为模型大小正在快速增长。因此，在本文中，我们将基于适配器的参数高效转移学习技术引入VL-BART和VL-T5等V＆L型号。我们在四个不同V＆L任务的统一多任务设置中评估我们的方法：VQAV2，GQA，NLVR2和MSCOCO图像标题。通过仔细的培训和彻底的实验，我们将三种流行的基于适配器的方法（适配器，Hyperformer，Compacter）基准，抵御标准的全部微调和最近提出的及时调整方法。我们还通过分享其权重以获得跨任务的知识来增强适配器的效率和性能。我们的结果表明，使用权重共享技术（总参数的4.4％）培训适配器可以匹配微调整个模型的性能。最后，我们提出了一个全面的分析，包括适配器和任务特定提示的组合以及V＆L对适配器进行培训的影响。我们的代码可用于：https://github.com/ylsung/vl_adapter。

Conventional fine-tuning encounters increasing difficulties given the size of current Pre-trained Language Models, which makes parameter-efficient tuning become the focal point of frontier research. Previous methods in this field add tunable adapters into MHA or/and FFN of Transformer blocks to enable PLMs achieve transferability. However, as an important part of Transformer architecture, the power of layer normalization for parameter-efficent tuning is ignored. In this paper, we first propose LN-tuning, by tuning the gain and bias term of Layer Normalization module with only 0.03\% parameters, which is of high time-efficency and significantly superior to baselines which are less than 0.1\% tunable parameters. Further, we study the unified framework of combining LN-tuning with previous ones and we find that: (1) the unified framework of combining prefix-tuning, the adapter-based method working on MHA, and LN-tuning achieves SOTA performance. (2) unified framework which tunes MHA and LayerNorm simultaneously can get performance improvement but those which tune FFN and LayerNorm simultaneous will cause performance decrease. Ablation study validates LN-tuning is of no abundant parameters and gives a further understanding of it.

情绪识别（ER）旨在将人的话语分类为不同的情感类别。基于本文和声学模式之间的早期融合和基于自我注意力的多模式相互作用，在本文中，我们提出了一种多模式多任务学习方法，用于从孤立的单个话语中进行ER。Iemocap基准测试的实验表明，我们提出的模型的表现要比我们对最新的改性的重新实现要好，并且比文献中所有其他单峰和多模式方法更好地实现了性能。此外，强大的基准和消融研究证明了我们提出的方法的有效性。我们在GitHub上公开提供所有代码。

Through solving pretext tasks, self-supervised learning leverages unlabeled data to extract useful latent representations replacing traditional input features in the downstream task. In audio/speech signal processing, a wide range of features where engineered through decades of research efforts. As it turns out, learning to predict such features (a.k.a pseudo-labels) has proven to be a particularly relevant pretext task, leading to useful self-supervised representations which prove to be effective for downstream tasks. However, methods and common practices for combining such pretext tasks for better performance on the downstream task have not been explored and understood properly. In fact, the process relies almost exclusively on a computationally heavy experimental procedure, which becomes intractable with the increase of the number of pretext tasks. This paper introduces a method to select a group of pretext tasks among a set of candidates. The method we propose estimates calibrated weights for the partial losses corresponding to the considered pretext tasks during the self-supervised training process. The experiments conducted on automatic speech recognition, speaker and emotion recognition validate our approach, as the groups selected and weighted with our method perform better than classic baselines, thus facilitating the selection and combination of relevant pseudo-labels for self-supervised representation learning.

端到端的语音到文本翻译模型通常使用预训练的语音编码器和预训练的文本解码器初始化。这导致了预训练和微调之间的显着训练差距，这在很大程度上是由于语音输出与解码器的文本输入之间的形式差异。在这项工作中，我们旨在弥合语音和文本之间的方式差距，以提高翻译质量。我们提出了一种基于变压器的新型模块M-Adapter，以使语音表示为文本。在缩小语音序列的同时，M-ADAPTER通过建模语音序列的全局和局部依赖性产生了对语音到文本翻译所需的特征。我们的实验结果表明，我们的模型在必要的基线上优于强大的基线，最高1个BLEU得分在必要时$ \ rightarrow $ de DataSet。\ footNote {我们的代码可在https://github.com/mingzi151/w2v2-v2-v2--proce上获得。英石。}

通过微调调整大型预训练模型（PTM）会施加过刺激的计算和存储负担。对参数有效调整（PET）的最新研究发现，与常规微调相比，仅优化以PTM为条件的一小部分参数才能产生PAR性能。通常，PET方法精确设计参数有效的模块（PET模块）可以应用于PTMS内部的任意细粒位置。但是，这些细粒度位置的有效性很大程度上依赖于复杂的手动指定，因此通常会产生次优的结果。与手动指定相反，我们以自动方式探索构建宠物模块。我们将自动\ textbf {s} earch \ textbf {s} parse \ textbf {s} \ textbf {p} arameter- \ textbf {e} fficbf {e} fficient \ textbf {t textbf {t} uning（s $^3 $ pet） 。基于各种PET方法的统一框架，S $^3 $ PET通过双层优化进行了可区分的PET结构搜索，并提出了移动的全局Sigmoid方法，以明确控制可训练的参数的数量。广泛的实验表明，S $^3 $ PET超过了具有较低训练参数的手册和随机结构。搜索结构可保留99 \％的微调性能，具有0.01 \％可训练的参数。此外，S $^3 $ PET的优势通过极低的训练参数预算（0.0009 \％$ \ sim $ 0.01 \％）进行扩增。搜索结构是可转移和解释的，为PET方法的未来设计提供了建议和指导。