Error correction is widely used in automatic speech recognition (ASR) to post-process the generated sentence, and can further reduce the word error rate (WER). Although multiple candidates are generated by an ASR system through beam search, current error correction approaches can only correct one sentence at a time, failing to leverage the voting effect from multiple candidates to better detect and correct error tokens. In this work, we propose FastCorrect 2, an error correction model that takes multiple ASR candidates as input for better correction accuracy. FastCorrect 2 adopts non-autoregressive generation for fast inference, which consists of an encoder that processes multiple source sentences and a decoder that generates the target sentence in parallel from the adjusted source sentence, where the adjustment is based on the predicted duration of each source token. However, there are some issues when handling multiple source sentences. First, it is non-trivial to leverage the voting effect from multiple source sentences since they usually vary in length. Thus, we propose a novel alignment algorithm to maximize the degree of token alignment among multiple sentences in terms of token and pronunciation similarity. Second, the decoder can only take one adjusted source sentence as input, while there are multiple source sentences. Thus, we develop a candidate predictor to detect the most suitable candidate for the decoder. Experiments on our inhouse dataset and AISHELL-1 show that FastCorrect 2 can further reduce the WER over the previous correction model with single candidate by 3.2% and 2.6%, demonstrating the effectiveness of leveraging multiple candidates in ASR error correction. FastCorrect 2 achieves better performance than the cascaded re-scoring and correction pipeline and can serve as a unified post-processing module for ASR.

Error correction techniques have been used to refine the output sentences from automatic speech recognition (ASR) models and achieve a lower word error rate (WER) than original ASR outputs. Previous works usually use a sequence-to-sequence model to correct an ASR output sentence autoregressively, which causes large latency and cannot be deployed in online ASR services. A straightforward solution to reduce latency, inspired by non-autoregressive (NAR) neural machine translation, is to use an NAR sequence generation model for ASR error correction, which, however, comes at the cost of significantly increased ASR error rate. In this paper, observing distinctive error patterns and correction operations (i.e., insertion, deletion, and substitution) in ASR, we propose FastCorrect, a novel NAR error correction model based on edit alignment. In training, FastCorrect aligns each source token from an ASR output sentence to the target tokens from the corresponding ground-truth sentence based on the edit distance between the source and target sentences, and extracts the number of target tokens corresponding to each source token during edition/correction, which is then used to train a length predictor and to adjust the source tokens to match the length of the target sentence for parallel generation. In inference, the token number predicted by the length predictor is used to adjust the source tokens for target sequence generation. Experiments on the public AISHELL-1 dataset and an internal industrial-scale ASR dataset show the effectiveness of FastCorrect for ASR error correction: 1) it speeds up the inference by 6-9 times and maintains the accuracy (8-14% WER reduction) compared with the autoregressive correction model; and 2) it outperforms the popular NAR models adopted in neural machine translation and text edition by a large margin.

Error correction in automatic speech recognition (ASR) aims to correct those incorrect words in sentences generated by ASR models. Since recent ASR models usually have low word error rate (WER), to avoid affecting originally correct tokens, error correction models should only modify incorrect words, and therefore detecting incorrect words is important for error correction. Previous works on error correction either implicitly detect error words through target-source attention or CTC (connectionist temporal classification) loss, or explicitly locate specific deletion/substitution/insertion errors. However, implicit error detection does not provide clear signal about which tokens are incorrect and explicit error detection suffers from low detection accuracy. In this paper, we propose SoftCorrect with a soft error detection mechanism to avoid the limitations of both explicit and implicit error detection. Specifically, we first detect whether a token is correct or not through a probability produced by a dedicatedly designed language model, and then design a constrained CTC loss that only duplicates the detected incorrect tokens to let the decoder focus on the correction of error tokens. Compared with implicit error detection with CTC loss, SoftCorrect provides explicit signal about which words are incorrect and thus does not need to duplicate every token but only incorrect tokens; compared with explicit error detection, SoftCorrect does not detect specific deletion/substitution/insertion errors but just leaves it to CTC loss. Experiments on AISHELL-1 and Aidatatang datasets show that SoftCorrect achieves 26.1% and 9.4% CER reduction respectively, outperforming previous works by a large margin, while still enjoying fast speed of parallel generation.

上下文偏见是端到端自动语音识别（ASR）系统的一项重要且具有挑战性现有方法主要包括上下文lm偏置，并将偏置编码器添加到端到端的ASR模型中。在这项工作中，我们介绍了一种新颖的方法，通过在端到端ASR系统之上添加上下文拼写校正模型来实现上下文偏见。我们将上下文信息与共享上下文编码器合并到序列到序列拼写校正模型中。我们提出的模型包括两种不同的机制：自动回旋（AR）和非自动回旋（NAR）。我们提出过滤算法来处理大尺寸的上下文列表以及性能平衡机制，以控制模型的偏置程度。我们证明所提出的模型是一种普遍的偏见解决方案，它是对域的不敏感的，可以在不同的情况下采用。实验表明，所提出的方法在ASR系统上的相对单词错误率（WER）降低多达51％，并且优于传统偏见方法。与AR溶液相比，提出的NAR模型可将模型尺寸降低43.2％，并将推断加速2.1倍。

连接派时间分类（CTC）的模型在自动语音识别（ASR）方面具有吸引力，因为它们的非自动性性质。为了利用仅文本数据，语言模型（LM）集成方法（例如重新纠正和浅融合）已被广泛用于CTC。但是，由于需要降低推理速度，因此他们失去了CTC的非自动性性本质。在这项研究中，我们提出了一种使用电话条件的蒙版LM（PC-MLM）的误差校正方法。在提出的方法中，掩盖了来自CTC的贪婪解码输出中的较不自信的单词令牌。然后，PC-MLM预测这些蒙版的单词令牌给定的单词和手机补充了CTC。我们进一步将其扩展到可删除的PC-MLM，以解决插入错误。由于CTC和PC-MLM均为非自动回旋模型，因此该方法可以快速LM集成。在域适应设置中对自发日本（CSJ）和TED-LIUM2语料库进行的实验评估表明，我们所提出的方法在推理速度方面优于重新逆转和浅融合，并且在CSJ上的识别准确性方面。

End-to-end speech recognition models trained using joint Connectionist Temporal Classification (CTC)-Attention loss have gained popularity recently. In these models, a non-autoregressive CTC decoder is often used at inference time due to its speed and simplicity. However, such models are hard to personalize because of their conditional independence assumption that prevents output tokens from previous time steps to influence future predictions. To tackle this, we propose a novel two-way approach that first biases the encoder with attention over a predefined list of rare long-tail and out-of-vocabulary (OOV) words and then uses dynamic boosting and phone alignment network during decoding to further bias the subword predictions. We evaluate our approach on open-source VoxPopuli and in-house medical datasets to showcase a 60% improvement in F1 score on domain-specific rare words over a strong CTC baseline.

误差校正技术仍然有效地通过自动语音识别（ASR）模型来完善输出。现有的端到端错误校正方法基于编码器架构架构过程在解码阶段中所有令牌，都会产生不良的延迟。在本文中，我们提出了一种利用校正操作预测的ASR误差校正方法。更具体地说，我们在编码器和解码器之间构建一个预测指标，以了解是否应保留一个令牌（“ k”），已删除（“ d”）或更改（“ C”）以限制解码仅为输入的一部分序列嵌入（“ C”令牌）用于快速推断。三个公共数据集的实验证明了拟议方法在减少ASR校正中解码过程的潜伏期中的有效性。与固体编码器基线相比，我们提出的两个模型的推理速度至少提高了3次（3.4次和5.7次），同时保持相同的准确性（分别降低0.53％和1.69％）。同时，我们生产并发布了为ASR错误校正社区做出贡献的基准数据集，以促进沿这一行的研究。

统一的流和非流式的双通（U2）用于语音识别的端到端模型在流传输能力，准确性，实时因素（RTF）和延迟方面表现出很大的性能。在本文中，我们呈现U2 ++，U2的增强版本，进一步提高了准确性。 U2 ++的核心思想是在训练中同时使用标签序列的前向和向后信息来学习更丰富的信息，并在解码时结合前向和后向预测以提供更准确的识别结果。我们还提出了一种名为SPECSUB的新数据增强方法，以帮助U2 ++模型更准确和强大。我们的实验表明，与U2相比，U2 ++在训练中显示了更快的收敛，更好地鲁棒性对解码方法，以及U2上的一致5 \％-8 \％字错误率降低增益。在Aishell-1的实验中，我们通过u2 ++实现了一个4.63 \％的字符错误率（cer），其中没有流媒体设置和5.05 \％，具有320ms延迟的流设置。据我们所知，5.05 \％是Aishell-1测试集上的最佳发布的流媒体结果。

Video dubbing aims to translate the original speech in a film or television program into the speech in a target language, which can be achieved with a cascaded system consisting of speech recognition, machine translation and speech synthesis. To ensure the translated speech to be well aligned with the corresponding video, the length/duration of the translated speech should be as close as possible to that of the original speech, which requires strict length control. Previous works usually control the number of words or characters generated by the machine translation model to be similar to the source sentence, without considering the isochronicity of speech as the speech duration of words/characters in different languages varies. In this paper, we propose a machine translation system tailored for the task of video dubbing, which directly considers the speech duration of each token in translation, to match the length of source and target speech. Specifically, we control the speech length of generated sentence by guiding the prediction of each word with the duration information, including the speech duration of itself as well as how much duration is left for the remaining words. We design experiments on four language directions (German -> English, Spanish -> English, Chinese <-> English), and the results show that the proposed method achieves better length control ability on the generated speech than baseline methods. To make up the lack of real-world datasets, we also construct a real-world test set collected from films to provide comprehensive evaluations on the video dubbing task.

自动语音识别（ASR）中编辑的后编辑需要自动纠正ASR系统产生的常见和系统错误。 ASR系统的输出在很大程度上容易出现语音和拼写错误。在本文中，我们建议使用强大的预训练的序列模型BART，BART进一步适应训练以作为剥夺模型，以纠正此类类型的错误。自适应培训是在通过合成诱导错误以及通过合并现有ASR系统中的实际错误获得的增强数据集上执行的。我们还提出了一种简单的方法，可以使用单词级别对齐来恢复输出。对重音语音数据的实验结果表明，我们的策略有效地纠正了大量的ASR错误，并在与竞争性基线相比时会产生改善的结果。我们还强调了在印地语语言中相关的语法误差校正任务中获得的负面结果，显示了通过我们建议的模型捕获更广泛上下文的限制。

虽然现代自动语音识别（ASR）系统可以实现高性能，但它们可能会产生削弱读者体验并对下游任务造成伤害的错误。为了提高ASR假设的准确性和可靠性，我们提出了一种用于语音识别器的跨模型后处理系统，其中1）熔断来自不同方式的声学特征和文本特征，2）接合置信度估计器和多个误差校正器任务学习时尚和3）统一纠错和话语抑制模块。与单模或单任务模型相比，我们提出的系统被证明更有效和高效。实验结果表明，我们的后处理系统导致对工业ASR系统的单扬声器和多扬声器语音相对降低的10％相对减少，每个令牌约为1.7ms延迟确保在流语音识别中可以接受后处理引入的额外延迟。

由于最近的自然语言处理的进步，几种作品已经将伯特的预先接受审查的屏蔽语言模型（MLM）应用于语音识别的后校正。然而，现有的预先训练的模型仅考虑语义校正，同时忽略了单词的语音特征。因此，语义后校正将降低性能，因为在中国ASR中同音误差相当常见。在本文中，我们提出了一种集体利用了语境化表示的新方法以及错误与其替换候选人之间的语音信息来缓解中国ASR的错误率。我们对现实世界语音识别数据集的实验结果表明，我们所提出的方法明显地低于基线模型的CER，其利用预先训练的BERT MLM作为校正器。

变形金刚最近在ASR领域主导。尽管能够产生良好的性能，但它们涉及自回归（AR）解码器，以一一生成令牌，这在计算上效率低下。为了加快推断，非自动回旋（NAR）方法，例如设计单步nar，以实现平行生成。但是，由于输出令牌内的独立性假设，单步nar的性能不如AR模型，尤其是在大规模语料库的情况下。改进单步nar面临两个挑战：首先，准确预测输出令牌的数量并提取隐藏的变量；其次，以增强输出令牌之间的相互依赖性建模。为了应对这两个挑战，我们提出了一个被称为Paraformer的快速准确的平行变压器。这利用了连续的基于集成和火的预测器来预测令牌的数量并生成隐藏的变量。然后，浏览语言模型（GLM）采样器会生成语义嵌入，以增强NAR解码器建模上下文相互依存的能力。最后，我们设计了一种策略来生成负面样本，以进行最小单词错误率训练以进一步提高性能。使用公共Aishell-1，Aishell-2基准和工业级别20,000小时任务的实验表明，拟议的Paraformer可以达到与最先进的AR变压器相当的性能，具有超过10倍的加速。

由于其误差传播，延迟较少和更少的参数较少的潜力，端到端语音到文本翻译〜（e2e-st）变得越来越受欢迎。鉴于三联培训语料库$ \ langle演讲，转录，翻译\ rangle $，传统的高质量E2E-ST系统利用$ \ langle演讲，转录\ rangle $配对预先培训模型，然后利用$ \ Langle演讲，翻译\ rangle $配对进一步优化它。然而，该过程仅涉及每个阶段的两个元组数据，并且该松散耦合不能完全利用三重态数据之间的关联。在本文中，我们试图基于语音输入模拟转录和翻译的联合概率，以直接利用这种三重态数据。基于此，我们提出了一种新的正规化方法，用于改进三重态数据中双路分解协议的模型培训，理论上应该是相等的。为实现这一目标，我们将两个Kullback-Leibler发散正规化术语介绍到模型培训目的中，以减少双路径输出概率之间的不匹配。然后，训练有素的模型可以通过预定义的早期停止标签自然地被视为E2E-ST模型。 Must-C基准测试的实验表明，我们所提出的方法在所有8个语言对上显着优于最先进的E2E-ST基线，同时在自动语音识别任务中实现更好的性能。我们的代码在https://github.com/duyichao/e2e -st-tda开放。

我们提出了一种用于计算自动语音识别（ASR）中错误率的新方法。这个新的指标是针对包含半字符的语言，可以以不同形式编写相同的字符。我们在印地语中实施了我们的方法论，这是指示上下文中的主要语言之一，我们认为这种方法可扩展到包含大型字符集的其他类似语言。我们称我们的指标替代单词错误率（AWER）和替代字符错误率（ACER）。我们使用wav2Vec 2.0 \ cite {baevski2020wav2vec}训练我们的ASR模型。此外，我们使用语言模型来改善我们的模型性能。我们的结果表明，在分析单词和角色级别的错误率方面有了显着提高，ASR系统的可解释性提高了高达$ 3 $ \％的AWER，印地语的ACER $ 7 $ \％。我们的实验表明，在具有复杂发音的语言中，有多种写单词而不改变其含义的方式。在这种情况下，Awer和Acer将更有用，而不是将其作为指标。此外，我们通过新的公制脚本为印地语开了一个21小时的新基准测试数据集。

The lack of label data is one of the significant bottlenecks for Chinese Spelling Check (CSC). Existing researches use the method of automatic generation by exploiting unlabeled data to expand the supervised corpus. However, there is a big gap between the real input scenario and automatic generated corpus. Thus, we develop a competitive general speller ECSpell which adopts the Error Consistent masking strategy to create data for pretraining. This error consistency masking strategy is used to specify the error types of automatically generated sentences which is consistent with real scene. The experimental result indicates our model outperforms previous state-of-the-art models on the general benchmark. Moreover, spellers often work within a particular domain in real life. Due to lots of uncommon domain terms, experiments on our built domain specific datasets show that general models perform terribly. Inspired by the common practice of input methods, we propose to add an alterable user dictionary to handle the zero-shot domain adaption problem. Specifically, we attach a User Dictionary guided inference module (UD) to a general token classification based speller. Our experiments demonstrate that ECSpell$^{UD}$, namely ECSpell combined with UD, surpasses all the other baselines largely, even approaching the performance on the general benchmark.

拼写错误纠正是自然语言处理中具有很长历史的主题之一。虽然以前的研究取得了显着的结果，但仍然存在挑战。在越南语中，任务的最先进的方法从其相邻音节中介绍了一个音节的上下文。然而，该方法的准确性可能是不令人满意的，因为如果模型可能会失去上下文，如果两个（或更多）拼写错误彼此静置。在本文中，我们提出了一种纠正越南拼写错误的新方法。我们使用深入学习模型解决错误错误和拼写错误错误的问题。特别地，嵌入层由字节对编码技术提供支持。基于变压器架构的序列模型的序列使我们的方法与上一个问题不同于同一问题的方法。在实验中，我们用大型合成数据集训练模型，这是随机引入的拼写错误。我们使用现实数据集测试所提出的方法的性能。此数据集包含11,202个以9,341不同的越南句子中的人造拼写错误。实验结果表明，我们的方法达到了令人鼓舞的表现，检测到86.8％的误差，81.5％纠正，分别提高了最先进的方法5.6％和2.2％。

最近，语音界正在看到从基于深神经网络的混合模型移动到自动语音识别（ASR）的端到端（E2E）建模的显着趋势。虽然E2E模型在大多数基准测试中实现最先进的，但在ASR精度方面，混合模型仍然在当前的大部分商业ASR系统中使用。有很多实际的因素会影响生产模型部署决定。传统的混合模型，用于数十年的生产优化，通常擅长这些因素。在不为所有这些因素提供优异的解决方案，E2E模型很难被广泛商业化。在本文中，我们将概述最近的E2E模型的进步，专注于解决行业视角的挑战技术。

我们提出了一种基于角色的非自由前口GEC方法，自动生成的字符变换。最近，校正编辑的每字分类已经证明了当前编码器解码器GEC系统有效，并行化替代方案。我们表明替换编辑可能是次优，导致形态学上丰富的语言中拼写，虚拟化和误差的规则爆炸，并提出了一种从GEC语料库产生字符变换的方法。最后，与宣传系统相比，我们培训捷克，德国和俄罗斯，达到固体成果和戏剧性加速的人物转型模型。源代码在https://github.com/ufal/wnut2021_character_transformations_gec发布。

End-to-end Speech Translation (E2E ST) aims to translate source speech into target translation without generating the intermediate transcript. However, existing approaches for E2E ST degrade considerably when only limited ST data are available. We observe that an ST model's performance strongly correlates with its embedding similarity from speech and transcript. In this paper, we propose Word-Aligned COntrastive learning (WACO), a novel method for few-shot speech-to-text translation. Our key idea is bridging word-level representations for both modalities via contrastive learning. We evaluate WACO and other methods on the MuST-C dataset, a widely used ST benchmark. Our experiments demonstrate that WACO outperforms the best baseline methods by 0.7-8.5 BLEU points with only 1-hour parallel data. Code is available at https://anonymous.4open.science/r/WACO .