口语语言理解（SLU）任务涉及从语音音频信号映射到语义标签。鉴于此类任务的复杂性，可能预期良好的性能需要大量标记的数据集，这很难为每个新任务和域收集。但是，最近的自我监督讲话表现的进步使得考虑使用有限标记的数据学习SLU模型是可行的。在这项工作中，我们专注于低资源讨论（ner）并解决问题：超越自我监督的预培训，我们如何使用未为任务注释的外部语音和/或文本数据？我们借鉴了各种方法，包括自我训练，知识蒸馏和转移学习，并考虑其对端到端模型和管道（语音识别后跟文本型号）的适用性。我们发现，这些方法中的几种方法可以在资源受限的环境中提高绩效，超出了训练有素的表示的福利。与事先工作相比，我们发现改进的F1分数高达16％。虽然最好的基线模型是一种管道方法，但使用外部数据时最终通过端到端模型实现的最佳性能。我们提供了详细的比较和分析，例如，端到端模型能够专注于更加立列人的单词。

通过共享数据集和基准，已经促进了语音处理的进展。历史上，这些都集中在自动语音识别（ASR），扬声器标识或其他较低级别的任务上。兴趣在更高层次的口语中越来越多，理解任务，包括使用端到端模型，但是此类任务的注释数据集较少。与此同时，最近的工作显示了预先培训通用表示的可能性，然后使用相对较少标记的数据进行微调的多个任务。我们建议为口语语言理解（屠宰）创建一套基准任务，由有限尺寸标记的培训集和相应的评估集组成。该资源将允许研究界跟踪进度，评估高级任务的预先接受预期的表示，并研究开放的问题，例如管道与端到端方法的实用性。我们介绍了雪橇基准套件的第一阶段，包括指定实体识别，情感分析和相应数据集上的ASR。我们专注于自然产生的（未读取或综合）语音和自由可用的数据集。我们为VoxceReb和Voxpopuli数据集的子集提供新的转录和注释，基线模型的评估指标和结果，以及重现基线的开源工具包，并评估新模型。

Spoken language understanding (SLU) tasks have been studied for many decades in the speech research community, but have not received as much attention as lower-level tasks like speech and speaker recognition. In particular, there are not nearly as many SLU task benchmarks, and many of the existing ones use data that is not freely available to all researchers. Recent work has begun to introduce such benchmark datasets for several tasks. In this work, we introduce several new annotated SLU benchmark tasks based on freely available speech data, which complement existing benchmarks and address gaps in the SLU evaluation landscape. We contribute four tasks: question answering and summarization involve inference over longer speech sequences; named entity localization addresses the speech-specific task of locating the targeted content in the signal; dialog act classification identifies the function of a given speech utterance. We follow the blueprint of the Spoken Language Understanding Evaluation (SLUE) benchmark suite. In order to facilitate the development of SLU models that leverage the success of pre-trained speech representations, we will be publishing for each task (i) annotations for a relatively small fine-tuning set, (ii) annotated development and test sets, and (iii) baseline models for easy reproducibility and comparisons. In this work, we present the details of data collection and annotation and the performance of the baseline models. We also perform sensitivity analysis of pipeline models' performance (speech recognizer + text model) to the speech recognition accuracy, using more than 20 state-of-the-art speech recognition models.

Collecting sufficient labeled data for spoken language understanding (SLU) is expensive and time-consuming. Recent studies achieved promising results by using pre-trained models in low-resource scenarios. Inspired by this, we aim to ask: which (if any) pre-training strategies can improve performance across SLU benchmarks? To answer this question, we employ four types of pre-trained models and their combinations for SLU. We leverage self-supervised speech and language models (LM) pre-trained on large quantities of unpaired data to extract strong speech and text representations. We also explore using supervised models pre-trained on larger external automatic speech recognition (ASR) or SLU corpora. We conduct extensive experiments on the SLU Evaluation (SLUE) benchmark and observe self-supervised pre-trained models to be more powerful, with pre-trained LM and speech models being most beneficial for the Sentiment Analysis and Named Entity Recognition task, respectively.

最近的言语和语言技术的方法预先rain非常大型模型，用于特定任务。然而，这种大型模型的好处通常仅限于世界上少数资源丰富的语言。在这项工作中，我们对来自印度次大陆的低资源语言构建ASR系统进行多种贡献。首先，我们从各种领域策划40个印度语言的17,000小时的原始语音数据，包括教育，新闻，技术和金融。其次，使用这种原始语音数据，我们预先存在于40个印度语言的Wav2Vec样式模型的多个变体。第三，我们分析佩带的模型以查找关键特点：码本矢量的类似探测音素在语言中共享，跨层的表示是语言系列的判别，并且注意力头通常会在小型本地窗口中注意。第四，我们微调了9种语言的下游ASR模型，并在3个公共数据集上获得最先进的结果，包括非常低的资源语言，如Sinhala和Nepali。我们的工作建立了多语言预介质是建立ASR系统的有效策略，为印度次大陆的语言上不同的扬声器建立ASR系统。

This paper describes a simple yet efficient repetition-based modular system for speeding up air-traffic controllers (ATCos) training. E.g., a human pilot is still required in EUROCONTROL's ESCAPE lite simulator (see https://www.eurocontrol.int/simulator/escape) during ATCo training. However, this need can be substituted by an automatic system that could act as a pilot. In this paper, we aim to develop and integrate a pseudo-pilot agent into the ATCo training pipeline by merging diverse artificial intelligence (AI) powered modules. The system understands the voice communications issued by the ATCo, and, in turn, it generates a spoken prompt that follows the pilot's phraseology to the initial communication. Our system mainly relies on open-source AI tools and air traffic control (ATC) databases, thus, proving its simplicity and ease of replicability. The overall pipeline is composed of the following: (1) a submodule that receives and pre-processes the input stream of raw audio, (2) an automatic speech recognition (ASR) system that transforms audio into a sequence of words; (3) a high-level ATC-related entity parser, which extracts relevant information from the communication, i.e., callsigns and commands, and finally, (4) a speech synthesizer submodule that generates responses based on the high-level ATC entities previously extracted. Overall, we show that this system could pave the way toward developing a real proof-of-concept pseudo-pilot system. Hence, speeding up the training of ATCos while drastically reducing its overall cost.

本文介绍了基于Wav2VEC 2.0的跨语言语音表示学习的大规模模型。我们在128种语言中培训最多2B个公共讲话音频的近半小时的型号的模型，比公共数据的数量级比最大的已知事先工作。我们的评估涵盖了广泛的任务，域，数据制度和语言，都是高低资源。在Covost-2语音翻译基准测试中，我们将先前的最先进的状态平均为7.4 BLEU超过21个翻译方向进入英语。对于语音识别，XLS-R在Babel，MLS，CommonVoice以及Voxpopuli上的最佳已知工作中提高，降低了相对的误差率14-34％。 XLS-R还在Voxlingua107语言识别上设置了新的技术状态。此外，我们表明，具有足够的模型规模，交叉思维预先预测可以在将英语演讲翻译成其他语言时才能优于英语撇印，这是一个有利于单晶的预借预制的设置。我们希望XLS-R可以帮助改善世界上更多语言的语音处理任务。

最近，蒙面的预测预训练在自我监督的学习（SSL）方面取得了显着的进展，以进行语音识别。它通常需要以无监督的方式获得的代码簿，从而使其准确和难以解释。我们提出了两种监督指导的代码书生成方法，以提高自动语音识别（ASR）的性能以及预训练效率，要么通过使用混合ASR系统来解码以生成音素级别对准（命名为PBERT），要么通过在上进行集群进行聚类。从端到端CTC模型（命名CTC聚类）提取的监督语音功能。混合动力和CTC模型均经过与微调相同的少量标记语音训练。实验表明，我们的方法对各种SSL和自我训练基准的优势具有显着优势，相对减少了17.0％。我们的预训练模型在非ASR语音任务中还显示出良好的可传递性。

我们总结了使用巨大的自动语音识别（ASR）模型的大量努力的结果，该模型使用包含大约一百万小时音频的大型，多样的未标记数据集进行了预训练。我们发现，即使对于拥有数万个小时的标记数据的非常大的任务，预训练，自我培训和扩大模型大小的组合也大大提高了数据效率。特别是，在具有34K小时标记数据的ASR任务上，通过微调80亿个参数预先训练的构象异构体模型，我们可以匹配最先进的（SOTA）性能（SOTA）的性能，只有3％的培训数据和通过完整的训练集可以显着改善SOTA。我们还报告了从使用大型预训练和自我训练的模型来完成一系列下游任务所获得的普遍利益，这些任务涵盖了广泛的语音域，并涵盖了多个数据集大小的大小，包括在许多人中获得SOTA性能公共基准。此外，我们利用预先训练的网络的学会表示，在非ASR任务上实现SOTA结果。

We study the capabilities of speech processing systems trained simply to predict large amounts of transcripts of audio on the internet. When scaled to 680,000 hours of multilingual and multitask supervision, the resulting models generalize well to standard benchmarks and are often competitive with prior fully supervised results but in a zero-shot transfer setting without the need for any fine-tuning. When compared to humans, the models approach their accuracy and robustness. We are releasing models and inference code to serve as a foundation for further work on robust speech processing.

在本文中，我们介绍了从包含超过80,000个小时的未标记的语音的大型数据集预处理捷克单语音频变压器方面的进展，随后使用内域数据组合对自动语音识别任务进行微调，并对模型进行微调。6000小时的跨域转录语音。我们在两个公共数据集（CommunVoice和Voxpopuli）和Malach Project中的一个非常具有挑战性的数据集中评估了各种微调设置的大量实验调色板。我们的结果表明，单语WAV2VEC 2.0模型是强大的ASR系统，它可以利用大型标记和未标记的数据集并成功与最先进的LVCSR系统竞争。此外，当没有用于目标ASR任务的培训数据时，WAV2VEC模型被证明是很好的零射门学习者。

Personal assistants, automatic speech recognizers and dialogue understanding systems are becoming more critical in our interconnected digital world. A clear example is air traffic control (ATC) communications. ATC aims at guiding aircraft and controlling the airspace in a safe and optimal manner. These voice-based dialogues are carried between an air traffic controller (ATCO) and pilots via very-high frequency radio channels. In order to incorporate these novel technologies into ATC (low-resource domain), large-scale annotated datasets are required to develop the data-driven AI systems. Two examples are automatic speech recognition (ASR) and natural language understanding (NLU). In this paper, we introduce the ATCO2 corpus, a dataset that aims at fostering research on the challenging ATC field, which has lagged behind due to lack of annotated data. The ATCO2 corpus covers 1) data collection and pre-processing, 2) pseudo-annotations of speech data, and 3) extraction of ATC-related named entities. The ATCO2 corpus is split into three subsets. 1) ATCO2-test-set corpus contains 4 hours of ATC speech with manual transcripts and a subset with gold annotations for named-entity recognition (callsign, command, value). 2) The ATCO2-PL-set corpus consists of 5281 hours of unlabeled ATC data enriched with automatic transcripts from an in-domain speech recognizer, contextual information, speaker turn information, signal-to-noise ratio estimate and English language detection score per sample. Both available for purchase through ELDA at http://catalog.elra.info/en-us/repository/browse/ELRA-S0484. 3) The ATCO2-test-set-1h corpus is a one-hour subset from the original test set corpus, that we are offering for free at https://www.atco2.org/data. We expect the ATCO2 corpus will foster research on robust ASR and NLU not only in the field of ATC communications but also in the general research community.

确保适当的标点符号和字母外壳是朝向应用复杂的自然语言处理算法的关键预处理步骤。这对于缺少标点符号和壳体的文本源，例如自动语音识别系统的原始输出。此外，简短的短信和微博的平台提供不可靠且经常错误的标点符号和套管。本调查概述了历史和最先进的技术，用于恢复标点符号和纠正单词套管。此外，突出了当前的挑战和研究方向。

Self-supervised pre-trained transformers have improved the state of the art on a variety of speech tasks. Due to the quadratic time and space complexity of self-attention, they usually operate at the level of relatively short (e.g., utterance) segments. In this paper, we study the use of context, i.e., surrounding segments, during fine-tuning and propose a new approach called context-aware fine-tuning. We attach a context module on top of the last layer of a pre-trained model to encode the whole segment into a context embedding vector which is then used as an additional feature for the final prediction. During the fine-tuning stage, we introduce an auxiliary loss that encourages this context embedding vector to be similar to context vectors of surrounding segments. This allows the model to make predictions without access to these surrounding segments at inference time and requires only a tiny overhead compared to standard fine-tuned models. We evaluate the proposed approach using the SLUE and Librilight benchmarks for several downstream tasks: Automatic speech recognition (ASR), named entity recognition (NER), and sentiment analysis (SA). The results show that context-aware fine-tuning not only outperforms a standard fine-tuning baseline but also rivals a strong context injection baseline that uses neighboring speech segments during inference.

开发语音技术是对低资源语言的挑战，其中注释和原始语音数据稀疏。马耳他是一种这样的语言。近年来，对马耳他的计算处理有所增加，包括语音技术，但后者的资源仍然稀疏。在本文中，我们考虑提高这些语言的语音识别的数据增强技术，专注于马耳他作为测试用例。我们考虑三种不同类型的数据增强：无监督的培训，多语言培训和合成演讲的使用作为培训数据。目标是确定这些技术或它们的组合，是改善起始点是大约7小时转录语音的语言的语言的最有效。我们的结果表明，在这里研究了三种数据增强技术，导致我们在不使用语言模型的情况下实现15％的绝对增长。

Self-supervised approaches for speech representation learning are challenged by three unique problems: (1) there are multiple sound units in each input utterance, (2) there is no lexicon of input sound units during the pre-training phase, and (3) sound units have variable lengths with no explicit segmentation. To deal with these three problems, we propose the Hidden-Unit BERT (HuBERT) approach for self-supervised speech representation learning, which utilizes an offline clustering step to provide aligned target labels for a BERT-like prediction loss. A key ingredient of our approach is applying the prediction loss over the masked regions only, which forces the model to learn a combined acoustic and language model over the continuous inputs. HuBERT relies primarily on the consistency of the unsupervised clustering step rather than the intrinsic quality of the assigned cluster labels. Starting with a simple k-means teacher of 100 clusters, and using two iterations of clustering, the HuBERT model either matches or improves upon the state-ofthe-art wav2vec 2.0 performance on the Librispeech (960h) and Libri-light (60,000h) benchmarks with 10min, 1h, 10h, 100h, and 960h fine-tuning subsets. Using a 1B parameter model, HuBERT shows up to 19% and 13% relative WER reduction on the more challenging dev-other and test-other evaluation subsets. 1

Modern speech recognition systems exhibits rapid performance degradation under domain shift. This issue is especially prevalent in data-scarce settings, such as low-resource languages, where diversity of training data is limited. In this work we propose M2DS2, a simple and sample-efficient finetuning strategy for large pretrained speech models, based on mixed source and target domain self-supervision. We find that including source domain self-supervision stabilizes training and avoids mode collapse of the latent representations. For evaluation, we collect HParl, a $120$ hour speech corpus for Greek, consisting of plenary sessions in the Greek Parliament. We merge HParl with two popular Greek corpora to create GREC-MD, a test-bed for multi-domain evaluation of Greek ASR systems. In our experiments we find that, while other Unsupervised Domain Adaptation baselines fail in this resource-constrained environment, M2DS2 yields significant improvements for cross-domain adaptation, even when a only a few hours of in-domain audio are available. When we relax the problem in a weakly supervised setting, we find that independent adaptation for audio using M2DS2 and language using simple LM augmentation techniques is particularly effective, yielding word error rates comparable to the fully supervised baselines.

无监督的语音识别表现出了使每种语言都可以访问的自动语音识别（ASR）系统的巨大潜力。但是，现有方法仍然严重依赖手工制作的预处理。与端到端进行监督语音识别的趋势类似，我们介绍了WAV2VEC-U 2.0，它消除了所有音频端的预处理，并通过更好的体系结构提高了准确性。此外，我们引入了一个辅助自我监督的目标，该目标将模型的预测与输入联系起来。实验表明，WAV2VEC-U 2.0在概念上更简单的同时，可以改善不同语言的无监督识别结果。

通过首先通过自动语音识别（ASR）转换话语，然后将输出馈送到基于文本的模型，通常通过转录语言理解（SLU）任务来解决。自我监督代表学习的最新进展旨在改善ASR组件。我们调查了是否对演讲的代表性学习已经成熟，以取代SLU中的ASR。我们将学位语音特征与Wav2Vec 2.0，最先进的ASR成绩单以及基于新型语音的名称实体识别任务的输入，是真实世界紧急呼叫和两个基于语音的命名实体识别任务的输入。现有的SLU基准。我们表明，学习的语音功能优于三种分类任务的ASR成绩单。对于机器翻译，ASR成绩单仍然是更好的选择。我们突出了Wav2VEC 2.0表示的内在稳健性，以失控的单词作为更好的性能的关键。

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。