Recently, there has been increasing progress in end-to-end automatic speech recognition (ASR) architecture, which transcribes speech to text without any pre-trained alignments. One popular end-to-end approach is the hybrid Connectionist Temporal Classification (CTC) and attention (CTC/attention) based ASR architecture, which utilizes the advantages of both CTC and attention. The hybrid CTC/attention ASR systems exhibit performance comparable to that of the conventional deep neural network (DNN) / hidden Markov model (HMM) ASR systems. However, how to deploy hybrid CTC/attention systems for online speech recognition is still a non-trivial problem. This paper describes our proposed online hybrid CTC/attention end-to-end ASR architecture, which replaces all the offline components of conventional CTC/attention ASR architecture with their corresponding streaming components. Firstly, we propose stable monotonic chunk-wise attention (sMoChA) to stream the conventional global attention, and further propose monotonic truncated attention (MTA) to simplify sMoChA and solve the training-and-decoding mismatch problem of sMoChA. Secondly, we propose truncated CTC (T-CTC) prefix score to stream CTC prefix score calculation. Thirdly, we design dynamic waiting joint decoding (DWJD) algorithm to dynamically collect the predictions of CTC and attention in an online manner. Finally, we use latency-controlled bidirectional long short-term memory (LC-BLSTM) to stream the widely-used offline bidirectional encoder network. Experiments with LibriSpeech English and HKUST Mandarin tasks demonstrate that, compared with the offline CTC/attention model, our proposed online CTC/attention model improves the real time factor in human-computer interaction services and maintains its performance with moderate degradation. To the best of our knowledge, this is the first work to provide the full-stack online solution for CTC/attention end-to-end ASR architecture.

When labeled data is insufficient, semi-supervised learning with the pseudo-labeling technique can significantly improve the performance of automatic speech recognition. However, pseudo-labels are often noisy, containing numerous incorrect tokens. Taking noisy labels as ground-truth in the loss function results in suboptimal performance. Previous works attempted to mitigate this issue by either filtering out the nosiest pseudo-labels or improving the overall quality of pseudo-labels. While these methods are effective to some extent, it is unrealistic to entirely eliminate incorrect tokens in pseudo-labels. In this work, we propose a novel framework named alternative pseudo-labeling to tackle the issue of noisy pseudo-labels from the perspective of the training objective. The framework comprises several components. Firstly, a generalized CTC loss function is introduced to handle noisy pseudo-labels by accepting alternative tokens in the positions of incorrect tokens. Applying this loss function in pseudo-labeling requires detecting incorrect tokens in the predicted pseudo-labels. In this work, we adopt a confidence-based error detection method that identifies the incorrect tokens by comparing their confidence scores with a given threshold, thus necessitating the confidence score to be discriminative. Hence, the second proposed technique is the contrastive CTC loss function that widens the confidence gap between the correctly and incorrectly predicted tokens, thereby improving the error detection ability. Additionally, obtaining satisfactory performance with confidence-based error detection typically requires extensive threshold tuning. Instead, we propose an automatic thresholding method that uses labeled data as a proxy for determining the threshold, thus saving the pain of manual tuning. Experiments demonstrate that alternative pseudo-labeling outperforms existing pseudo-labeling approaches on datasets in various domains and languages.

Deep learning is widely used for target detection and direction-of-arrival (DOA) estimation due to its powerful data fitting capability. However, limited by different environments and number of sound sources, it is difficult to be applied to complex underwater environments. We propose a two-stage approach called Beam-Network for underwater acoustic DOA estimation. In the first stage, local beam patterns with different data augmentation methods, carrying the essential information required for target detection, are used as the input feature to our model. In the second stage, an adaptive convolutional neural network (CNN) is proposed to construct a classification model. Only single-source data are required for model training and data from multi sources can be tested. What’s more, the model is suitable for arrays with different numbers of hydrophones in different geometrical arrangements. The performance of the proposed method is evaluated by comparing with mainstream DOA estimation algorithms such as CBF, MUSIC, MVDR, and SBL. In three simulation scenarios and two sets of recorded data from different marine environments, the proposed method has higher directivity and lower angular root-mean-squared error (RMSE).

Active noise control (ANC) systems are widely used to cancel unwanted noise. However, for high-level noise, the residual error signal cannot be fully eliminated because of the nonlinearity of the secondary path, resulting in the diverging of the adaptive filter. In this letter, we propose a secondary path-decoupled ANC (SPD-ANC) algorithm based on deep learning. Specifically, the secondary path decoupled module consisting of two time-domain convolutional recurrent networks, one for modeling the nonlinear secondary path and the other for modeling the reverse process, is employed to calculate the secondary path-decoupled (SPD) error signal. The control signal is then generated by an adaptive filter that is optimized towards minimizing the SPD error signal. Simulation results indicate that the proposed method outperforms the conventional ANC methods under different conditions.

The cross-domain performance of automatic speech recognition (ASR) could be severely hampered due to the mismatch between training and testing distributions. Since the target domain usually lacks labeled data, and domain shifts exist at acoustic and linguistic levels, it is challenging to perform unsupervised domain adaptation (UDA) for ASR. Previous work has shown that self-supervised learning (SSL) or pseudo-labeling (PL) is effective in UDA by exploiting the self-supervisions of unlabeled data. However, these self-supervisions also face performance degradation in mismatched domain distributions, which previous work fails to address. This work presents a systematic UDA framework to fully utilize the unlabeled data with self-supervision in the pre-training and fine-tuning paradigm. On the one hand, we apply continued pre-training and data replay techniques to mitigate the domain mismatch of the SSL pre-trained model. On the other hand, we propose a domain-adaptive fine-tuning approach based on the PL technique with three unique modifications: Firstly, we design a dual-branch PL method to decrease the sensitivity to the erroneous pseudo-labels; Secondly, we devise an uncertainty-aware confidence filtering strategy to improve pseudo-label correctness; Thirdly, we introduce a two-step PL approach to incorporate target domain linguistic knowledge, thus generating more accurate target domain pseudo-labels. Experimental results on various cross-domain scenarios demonstrate that the proposed approach effectively boosts the cross-domain performance and significantly outperforms previous approaches.

Recently, we have witnessed excellent improvement of end-to-end (E2E) automatic speech recognition (ASR). However, how to tackle the long-tailed data distribution problem while maintaining E2E ASR models' performance for high-frequency tokens is still challenging. To solve this challenge, we propose a novel decoupled ASR learning method for the sequence-to-sequence ASR architecture in this paper. Our method decouples the learning procedure of this model into two stages: representation learning and classification learning. In the representation learning stage, we use the encoder output of a pretrained language model as one of the ASR model’s learning targets, and propose threshold log cosine embedding loss (TLCE-loss) as the objective function. A frequency-mask cross-entropy loss (FMCE-loss) is also designed as an auxiliary loss. In the classification learning stage, we find that introducing a temperature into softmax function helps reduce the influence of negative samples on tail classes, thus mitigating the biased learning process for the classifier. Furthermore, we propose a weighted softmax (w-softmax) to adjust ASR posterior probabilities according to the token appearing frequency during inference. Additionally, we introduce tail word/character error rate (TWER / TCER) and head word/character error rate (HWER / HCER) that respectively evaluate the ASR accuracy for tail and head words/characters. Experimental results on the Switchboard and HKUST corpora show that our proposed method greatly outperforms the baseline, especially in TWER / TCER reduction. To the best of our knowledge, this is the first work to use a decoupled ASR learning method to alleviate the long-tailed problem in sequence-to-sequence ASR.