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。

无监督的文本到语音综合（TTS）系统学会通过观察以下语言来生成与任何语言中任何书面句子相对应的语音波形：1）用该语言收集的未转录语音波形的集合； 2）用该语言编写的文本集合，无需访问任何抄录的语音。开发这种系统可以显着提高语言技术对语言的可用性，而无需大量平行的语音和文本数据。本文提出了一个基于对齐模块的无监督的TTS系统，该模块输出了伪文本和另一个使用伪文本进行训练和真实文本进行推理的合成模块。我们的无监督系统可以以七种语言的方式实现与监督系统相当的性能，每种语音约10-20小时。还对文本单元和声码器的效果进行了仔细的研究，以更好地了解哪些因素可能影响无监督的TTS性能。可以在https://cactuswiththoughts.github.io/unsuptts-demo上找到我们的模型生成的样品，可以在https://github.com/lwang114/unsuptts上找到我们的代码。

This paper describes the ESPnet Unsupervised ASR Open-source Toolkit (EURO), an end-to-end open-source toolkit for unsupervised automatic speech recognition (UASR). EURO adopts the state-of-the-art UASR learning method introduced by the Wav2vec-U, originally implemented at FAIRSEQ, which leverages self-supervised speech representations and adversarial training. In addition to wav2vec2, EURO extends the functionality and promotes reproducibility for UASR tasks by integrating S3PRL and k2, resulting in flexible frontends from 27 self-supervised models and various graph-based decoding strategies. EURO is implemented in ESPnet and follows its unified pipeline to provide UASR recipes with a complete setup. This improves the pipeline's efficiency and allows EURO to be easily applied to existing datasets in ESPnet. Extensive experiments on three mainstream self-supervised models demonstrate the toolkit's effectiveness and achieve state-of-the-art UASR performance on TIMIT and LibriSpeech datasets. EURO will be publicly available at https://github.com/espnet/espnet, aiming to promote this exciting and emerging research area based on UASR through open-source activity.

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

我们介绍了一种用于跨语言训练ASR系统的方法，使用目标语言绝对没有转录的训练数据，并且没有相关语言的语音知识。我们的方法使用了一种解密算法的新应用，该算法仅在目标语言中仅操作不配对的语音和文本数据。我们将此破译应用于由通用电话识别器产生的电话序列，由语言语音语料库培训，我们遵循平稳半监督培训，以获得新语言的声学模型。据我们所知，这是零资源交叉语言ASR的第一种实用方法，不依赖于任何手工制作的语音信息。我们对来自Globalphone语料库的读语音进行了实验，并表明可以在目标语言中仅在20分钟的数据上学习解密模型。当用于生成半监督培训的伪标签时，我们获得了比在同一数据上培训的等同完全监督模型的25％至仅5％的绝对差。

最近已显示ASR最近取得了出色的性能。但是，他们中的大多数都依靠大量的配对数据，这对于全球低资源语言是不可行的。本文研究了如何直接从未配对的电话序列和语音话语中学习。我们设计了两个阶段的迭代框架。 GAN培训在第一阶段被采用，以找到未配对的语音和电话序列之间的映射关系。在第二阶段，引入了另一个HMM模型以从发电机的输出中训练，从而提高了性能，并为下一次迭代提供了更好的细分。在实验中，我们首先研究模型设计的不同选择。然后，我们将框架与不同类型的基准进行比较：（i）受监督的方法（ii）基于声学单元发现的方法（III）方法从未配对的数据中学习。我们的框架的表现始终如一，比所有基于TIMIT数据集从未配对数据中学习的所有声学单元发现方法和以前的方法更好。

我们介绍了一种无线文字语音转换（S2ST）系统，可以将来自一种语言的语音转换为另一种语言，并且可以在不需要任何文本数据的情况下构建。与文献中的现有工作不同，我们解决了模拟多扬声器目标语音的挑战，并用现实世界的S2ST数据训练系统。我们方法的关键是一种自我监督的单位语音标准化技术，该标准化技术将预先训练的语音编码器具有来自多个扬声器的配对声音，以及单个参考扬声器，以减少由于复印件引起的变化，同时保留词汇内容。只有10分钟的语音标准化的配对数据，我们在培训\ vp〜s2st数据集上的S2ST模型时获得平均3.2 BLEU增益，而不是在未标准化的语音目标上培训的基线。我们还将自动开采的S2ST数据纳入并显示额外的2.0 BLEU增益。据我们所知，我们是第一个建立无线的S2ST技术，可以用真实世界的数据培训，并为多种语言配对工作。

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。

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.

语音的视频录制包含相关的音频和视觉信息，为语音表示从扬声器的唇部运动和产生的声音提供了强大的信号。我们介绍了视听隐藏单元BERT（AV-HUBERT），是视听语音的自我监督的代表学习框架，这些屏幕屏蔽了多流视频输入并预测自动发现和迭代地精制多模式隐藏单元。 AV-HUBERT学习强大的视听语音表示，这些语音表示受益于唇读和自动语音识别。在最大的公众唇读基准LRS3（433小时）中，AV-Hubert达到32.5％WER，只有30个小时的标签数据，优于前一种最先进的方法（33.6％）培训，达到了一千次转录的视频数据（31k小时）。当使用来自LRS3的所有433小时的标记数据并结合自培训时，唇读WER进一步降低至26.9％。使用我们在相同的基准测试中使用您的视听表示，用于音频语音识别的相对效率为40％，而最先进的性能（1.3％Vs 2.3％）。我们的代码和模型可在https://github.com/facebookResearch/av_hubert获得

We show for the first time that learning powerful representations from speech audio alone followed by fine-tuning on transcribed speech can outperform the best semi-supervised methods while being conceptually simpler. wav2vec 2.0 masks the speech input in the latent space and solves a contrastive task defined over a quantization of the latent representations which are jointly learned. Experiments using all labeled data of Librispeech achieve 1.8/3.3 WER on the clean/other test sets. When lowering the amount of labeled data to one hour, wav2vec 2.0 outperforms the previous state of the art on the 100 hour subset while using 100 times less labeled data. Using just ten minutes of labeled data and pre-training on 53k hours of unlabeled data still achieves 4.8/8.2 WER. This demonstrates the feasibility of speech recognition with limited amounts of labeled data. 1 1 Code and models are available at https://github.com/pytorch/fairseq Preprint. Under review.

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。

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.

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。

最新的语音识别模型依赖于大型监督数据集，这些数据集对于许多低资源语言而言不可用。在这项工作中，我们提出了一条语音识别管道，该管道不需要目标语言的任何音频。唯一的假设是我们可以访问原始文本数据集或一组N-Gram统计信息。我们的语音管道包括三个组成部分：声学，发音和语言模型。与标准管道不同，我们的声学和​​发音模型在没有任何监督的情况下使用多语言模型。语言模型是使用n-gram统计信息或原始文本数据集构建的。我们通过将其与Croubadan结合使用：一种大型濒危语言N-Gram数据库来构建1909年语言的语音识别。此外，我们在两个数据集中测试了129种语言的方法：常见语音和CMU Wilderness数据集。我们在使用Crubadan统计数据的荒野数据集上获得了50％的CER和74％WER，并在使用10000原始文本说话时将其提高到45％的CER和69％。

In this paper, we propose a novel multi-modal multi-task encoder-decoder pre-training framework (MMSpeech) for Mandarin automatic speech recognition (ASR), which employs both unlabeled speech and text data. The main difficulty in speech-text joint pre-training comes from the significant difference between speech and text modalities, especially for Mandarin speech and text. Unlike English and other languages with an alphabetic writing system, Mandarin uses an ideographic writing system where character and sound are not tightly mapped to one another. Therefore, we propose to introduce the phoneme modality into pre-training, which can help capture modality-invariant information between Mandarin speech and text. Specifically, we employ a multi-task learning framework including five self-supervised and supervised tasks with speech and text data. For end-to-end pre-training, we introduce self-supervised speech-to-pseudo-codes (S2C) and phoneme-to-text (P2T) tasks utilizing unlabeled speech and text data, where speech-pseudo-codes pairs and phoneme-text pairs are a supplement to the supervised speech-text pairs. To train the encoder to learn better speech representation, we introduce self-supervised masked speech prediction (MSP) and supervised phoneme prediction (PP) tasks to learn to map speech into phonemes. Besides, we directly add the downstream supervised speech-to-text (S2T) task into the pre-training process, which can further improve the pre-training performance and achieve better recognition results even without fine-tuning. Experiments on AISHELL-1 show that our proposed method achieves state-of-the-art performance, with a more than 40% relative improvement compared with other pre-training methods.

本文介绍了Thuee团队的语音识别系统，用于IARPA Open自动语音识别挑战（OpenASR21），并进行了进一步的实验探索。我们在受限和受约束的训练条件下取得了出色的成果。对于受限的训练条件，我们基于标准混合体系结构构建基本ASR系统。为了减轻摄影库（OOV）的问题，我们使用针对OOV和潜在的新单词的素式至phoneme（G2P）技术扩展了发音词典。采用了标准的声学模型结构，例如CNN-TDNN-F和CNN-TDNN-F-A。此外，还应用了多种数据增强技术。对于约束训练条件，我们使用自我监督的学习框架WAV2VEC2.0。我们在公开可用的预训练XLSR-53的基础上使用连接式时间分类（CTC）标准进行各种微调技术。我们发现，在将WAV2VEC2.0预训练的模型应用于基于编码器的CTC/CTC/COATION ASR体系结构时，前端特征提取器在将WAV2VEC2.0预训练的模型应用时起着重要作用。通过将目标语言用作为前端功能提取器使用的CTC模型填充可以实现额外的改进。

In speech recognition, it is essential to model the phonetic content of the input signal while discarding irrelevant factors such as speaker variations and noise, which is challenging in low-resource settings. Self-supervised pre-training has been proposed as a way to improve both supervised and unsupervised speech recognition, including frame-level feature representations and Acoustic Word Embeddings (AWE) for variable-length segments. However, self-supervised models alone cannot learn perfect separation of the linguistic content as they are trained to optimize indirect objectives. In this work, we experiment with different pre-trained self-supervised features as input to AWE models and show that they work best within a supervised framework. Models trained on English can be transferred to other languages with no adaptation and outperform self-supervised models trained solely on the target languages.

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。

While the Turkish language is listed among low-resource languages, literature on Turkish automatic speech recognition (ASR) is relatively old. In this report, we present our findings on Turkish ASR with speech representation learning using HUBERT. We investigate pre-training HUBERT for Turkish with large-scale data curated from online resources. We pre-train our model using 6,500 hours of speech data from YouTube. The results show that the models are not ready for commercial use since they are not robust against disturbances that typically occur in real-world settings such as variations in accents, slang, background noise and interference. We analyze typical errors and the limitations of the models for use in commercial settings.