26 accepted papers
Word error rate (WER) estimation aims to evaluate the quality of an automatic speech recognition (ASR) system’s output without requiring ground-truth labels. This task has gained increasing attention as advanced ASR systems are trained on large amounts of data. In this context, the computational eff…
Word error rate (WER) is a metric used to evaluate the quality of transcriptions produced by Automatic Speech Recognition (ASR) systems. In many applications, it is of interest to estimate WER given a pair of a speech utterance and a transcript. Previous work on WER estimation focused on building mo…
Separation of overlapping speakers remains an active area of speech technology research. Many deep neural network (DNN) separation models propose modelling local and global temporal context separately using alternating DNN layers. Two such models are SepFormer and TD-Conformer. The largest configura…
Code-switching (CS) is the process of speakers interchanging between two or more languages which in the modern world becomes increasingly common. In order to better describe CS speech the Matrix Language Frame (MLF) theory introduces the concept of a Matrix Language, which is the language that provi…
Neural network based approaches to speech enhancement have shown to be particularly powerful, being able to leverage a data-driven approach to result in a significant performance gain versus other approaches. Such approaches are reliant on artificially created labelled training data such that the ne…
Neural networks have been successfully used for non-intrusive speech intelligibility prediction. Recently, the use of feature representations sourced from intermediate layers of pre-trained self-supervised and weakly-supervised models has been found to be particularly useful for this task. This work…
In Automatic Speech Recognition (ASR), teacher-student (T/S) training has shown to perform well for domain adaptation with small amount of training data. However, adaption without ground-truth labels is still challenging. A previous study has shown the effectiveness of using ensemble teacher models…
There is a growing interest in cost-effective self-supervised fine-tuning (SSFT) of self-supervised learning (SSL)-based speech models to obtain task-specific representations. These task-specific representations are used for robust performance on various downstream tasks by fine-tuning on the labell…
Speech separation models are used for isolating individual speakers in many speech processing applications. Deep learning models have been shown to lead to state-of-the-art (SOTA) results on a number of speech separation benchmarks. One such class of models known as temporal convolutional networks (…
Recent work in the domain of speech enhancement has explored the use of self-supervised speech representations to aid in the training of neural speech enhancement models. However, much of this work focuses on using the deepest or final outputs of self supervised speech representation models, rather…
Knowledge distillation (KD) has widely been used for model compression and domain adaptation for speech applications. In the presence of multiple teachers, knowledge can easily be transferred to the student by averaging the models output. However, previous research shows that the student do not adap…
In pronunciation assessment, the assessor’s perception is influenced by a particular pronunciation template. This assessor may hold a bias towards certain variations in pronunciation which do not necessarily impact communication, yet they may be penalized during the assessment. This work proposes a…
This paper proposes an unsupervised data selection method by using a submodular function based on contrastive loss ratios of target and training data sets. A model using a contrastive loss function is trained on both sets. Then the ratio of frame-level losses for each model is used by a submodular f…
Anomalous audio in speech recordings is often caused by speaker voice distortion, external noise, or even electric interferences. These obstacles have become a serious problem in some fields, such as high-quality dubbing and speech processing. In this paper, a novel approach using a temporal convolu…
This paper proposes an adaptation method for end-to-end speech recognition. In this method, multiple automatic speech recognition (ASR) 1-best hypotheses are integrated in the computation of the connectionist temporal classification (CTC) loss function. The integration of multiple ASR hypotheses hel…
Many-to-many voice conversion with non-parallel training data has seen significant progress in recent years. It is challenging because of lacking of ground truth parallel data. StarGAN-based models have gained attentions because of their efficiency and effective. However, most of the StarGAN-based w…
In this paper, a hierarchical attention network is proposed to generate utterance-level embeddings (H-vectors) for speaker identification and verification. Since different parts of an utterance may have different contributions to speaker identities, the use of hierarchical structure aims to learn sp…
In earlier work we studied the effect of statistical normalisation for phase-based features and observed it leads to a significant robustness improvement. This paper explores the extension of the generalised Vector Taylor Series (gVTS) noise compensation approach to the group delay (GD) domain. We d…
Speaker diarisation addresses the question of “who speaks when” in audio recordings, and has been studied extensively in the context of tasks such as broadcast news, meetings, etc. Performing diarisation on individual headset microphone (IHM) channels is sometimes assumed to easily give the desired…
This paper introduces the development of ShefCE: a Cantonese-English bilingual speech corpus from L2 English speakers in Hong Kong. Bilingual parallel recording materials were chosen from TED online lectures. Script selection were carried out according to bilingual consistency (evaluated using a mac…
In earlier work we have proposed a source-filter decomposition of speech through phase-based processing. The decomposition leads to novel speech features that are extracted from the filter component of the phase spectrum. This paper analyses this spectrum and the proposed representation by evaluatin…
Quality estimation models are used to predict the quality of the output from a spoken language translation (SLT) system. When these scores are used to rerank a k-best list, the rank of the scores is more important than their absolute values. This paper proposes groupwise learning to model this rank.…
High performance diarisation is a necessity for a variety of applications, and the task has been studied extensively in the context of broadcast news and meeting processing. Upon introduction of the task in NIST led evaluations, diarisation error rate (DER) was introduced as the standard metric for…
Deep Neural Network (DNN) has become a standard method in many ASR tasks. Recently there is considerable interest in “informed training” of DNNs, where DNN input is augmented with auxiliary codes, such as i-vectors, speaker codes, speaker separation bottleneck (SSBN) features, etc. This paper compar…
This paper presents a novel system for automatic assessment of pronunciation quality of English learner speech, based on deep neural network (DNN) features and phoneme specific discriminative classifiers. DNNs trained on a large corpus of native and non-native learner speech are used to extract phon…
Spoken language translation (SLT) combines automatic speech recognition (ASR) and machine translation (MT). During the decoding stage, the best hypothesis produced by the ASR system may not be the best input candidate to the MT system, but making use of multiple sub-optimal ASR results in SLT has be…