Self-Supervised Learning (SSL) has gained traction for its ability to learn rich representations with low labeling costs, applicable across diverse downstream tasks. However, assessing the downstream-task performance remains challenging due to the cost of extra training and evaluation. Existing methods for task-agnostic evaluation also require extra training or hyper-parameter tuning. We propose a novel evaluation metric using large language models (LLMs). By inputting discrete token sequences and minimal domain cues derived from SSL models into LLMs, we obtain the mean log-likelihood; these cues guide in-context learning, rendering the score more reliable without extra training or hyperparameter tuning. Experimental results show a correlation between LLM-based scores and automatic speech recognition task. Additionally, our findings reveal that LLMs not only functions as an SSL evaluation tools but also provides inference-time embeddings that are useful for speaker verification task.

Speech signals, typically sampled at rates in the tens of thousands per second, contain redundancies, evoking inefficiencies in sequence modeling. High-dimensional speech features such as spectrograms are often used as the input for the subsequent model. However, they can still be redundant. Recent investigations proposed the use of discrete speech units derived from self-supervised learning representations, which significantly compresses the size of speech data. Applying various methods, such as de-duplication and subword modeling, can further compress the speech sequence length. Hence, training time is significantly reduced while retaining notable performance. In this study, we undertake a comprehensive and systematic exploration into the application of discrete units within end-to-end speech processing models. Experiments on 12 automatic speech recognition, 3 speech translation, and 1 spoken language understanding corpora demonstrate that discrete units achieve reasonably good results in almost all the settings. Our configurations and trained models are released in ESPnet to foster future research efforts.

Recently, the usefulness of self-supervised representation learning (SSRL) methods has been confirmed in various downstream tasks. Many of these models, as exemplified by HuBERT and WavLM, use pseudo-labels generated from spectral features or the model’s own representation features. From previous studies, it is known that the pseudo-labels contain semantic information. However, the masked prediction task, the learning criterion of HuBERT, focuses on local contextual information and may not make effective use of global semantic information such as speaker, theme of speech, and so on. In this paper, we propose a new approach to enrich the semantic representation of HuBERT. We apply topic model to pseudo-labels to generate a topic label for each utterance. An auxiliary topic classification task is added to HuBERT by using topic labels as teachers. This allows additional global semantic information to be incorporated in an unsupervised manner. Experimental results demonstrate that our method achieves comparable or better performance than the baseline in most tasks, including automatic speech recognition and five out of the eight SUPERB tasks. Moreover, we find that topic labels include various information about utterance, such as gender, speaker, and its theme. This highlights the effectiveness of our approach in capturing multifaceted semantic nuances.

Distant Automatic Speech Recognition (DASR) stands as a crucial aspect in the realm of speech and audio processing. Recent advancements have spotlighted the efficacy of pre-trained speech foundation models, exemplified by Whisper, garnering considerable attention in the speech-processing domain. These models, trained on hundreds of thousands of hours of speech data, exhibit notable strengths in performance and generalization across various zero-shot scenarios. However, a limitation arises from their exclusive handling of single-channel input due to challenges in accumulating extensive multi-channel speech data. The spatial information in the multi-channel input is important for the DASR task. This study introduces an innovation by enabling the incorporation of multi-channel (MC) signals into the pre-trained Whisper model, called MC-Whisper. The proposed model introduces a multi-channel speech processing branch as a sidecar, to maximize the utilization of the foundation model's ability to handle multi-channel input. Experimental results on the distant microphone speech recordings from AMI meeting corpus demonstrate substantial improvements through the proposed approach.

Non-autoregressive (NAR) models for automatic speech recognition (ASR) aim to achieve high accuracy and fast inference by simplifying the autoregressive (AR) generation process of conventional models. Connectionist temporal classification (CTC) is one of the key techniques used in NAR ASR models. In this paper, we propose a new model combining CTC and a latent variable model, which is one of the state-of-the-art models in the neural machine translation research field. A new neural network architecture and formulation specialized for ASR application are introduced. In the proposed model, CTC alignment is assumed to be dependent on the latent variables that are expected to capture dependencies between tokens. Experimental results on a 100 hours subset of Librispeech corpus showed the best recognition accuracy among CTC-based NAR models. On the TED-LIUM2 corpus, the best recognition accuracy is achieved including AR E2E models with faster inference speed.

Unsupervised topic clustering of spoken audio is an important research topic for zero-resourced unwritten languages. A classical approach is to find a set of spoken terms from only the audio based on dynamic time warping or generative modeling (e.g., hidden Markov model), and apply a topic model to classify topics. The spoken term discovery is the most important and difficult part. In this paper, we propose to combine self-supervised representation learning (SSRL) methods as a component of spoken term discovery and probabilistic topic models. Most SSRL methods pre-train a model which predicts high-quality pseudo labels generated from an audio-only corpus. These pseudo labels can be used to produce a sequence of pseudo subwords by applying deduplication and a subword model. Then, we apply a topic model based on latent Dirichlet allocation for these pseudo-subword sequences in an unsupervised manner. The clustering performance is evaluated on the Fisher corpus using normalized mutual information. We confirm the improvement of the proposed method and its effectiveness compared to an existing approach using dynamic time warping and topic models although the experimental setups are not directly comparable.

Self-supervised learning (SSL) has become one of the most important technologies to realize spoken dialogue systems for languages that do not have much audio data and its transcription available. Speech representation models are one of the keys to achieving this, and have been actively studied in recent years. Among them, Hidden-Unit BERT (HuBERT) has shown promising results in automatic speech recognition (ASR) tasks. However, previous studies have investigated with limited iterations and cluster units. We explore HuBERT with larger numbers of clusters and iterations in order to obtain better speech representation. Furthermore, we introduce the Bayesian Information Criterion (BIC) as the performance measure of the model. Experimental results show that our model achieves the best performance in 5 out of 8 scores in the 4 metrics for the Zero Resource Speech 2021 task. It also outperforms the HuBERT BASE model trained with 960-hour LibriSpeech (LS) even though our model is only trained with 100-hour LS. In addition, we report that BIC is useful as a clue for determining the appropriate number of clusters to improve performance on phonetic, lexical, and syntactic metrics. Finally, we show that these findings are also effective for the ASR task.

This work presents our end-to-end (E2E) automatic speech recognition (ASR) model targetting at robust speech recognition, called Integraded speech Recognition with enhanced speech Input for Self-supervised learning representation (IRIS).Compared with conventional E2E ASR models, the proposed E2E model integrates two important modules including a speech enhancement (SE) module and a self-supervised learning representation (SSLR) module.The SE module enhances the noisy speech.Then the SSLR module extracts features from enhanced speech to be used for speech recognition (ASR).To train the proposed model, we establish an efficient learning scheme.Evaluation results on the monaural CHiME-4 task show that the IRIS model achieves the best performance reported in the literature for the single-channel CHiME-4 benchmark (2.0% for the real development and 3.9% for the real test) thanks to the powerful pre-trained SSLR module and the finetuned SE module.

Self-supervised pretraining on speech data has achieved a lot of progress. High-fidelity representation of the speech signal is learned from a lot of untranscribed data and shows promising performance. Recently, there are several works focusing on evaluating the quality of self-supervised pretrained representations on various tasks with-out domain restriction, e.g. SUPERB. However, such evaluations do not provide a comprehensive comparison among many ASR benchmark corpora. In this paper, we focus on the general applications of pretrained speech representations, on advanced end-to-end automatic speech recognition (E2E-ASR) models. We select sev-eral pretrained speech representations and present the experimental results on various open-source and publicly available corpora for E2E-ASR. Without any modification of the back-end model archi-tectures or training strategy, some of the experiments with pretrained representations, e.g., WSJ, WSJ0-2mix with HuBERT, reach or out-perform current state-of-the-art (SOTA) recognition performance. Moreover, we further explore more scenarios for whether the pre-training representations are effective, such as the cross-language or overlapped speech. The scripts, configuratons and the trained mod-els have been released in ESPnet to let the community reproduce our experiments and improve them.

We present a system for the Zero Resource Speech Challenge 2021, which combines a Contrastive Predictive Coding (CPC) with deep cluster. In deep cluster, we first prepare pseudo-labels obtained by clustering the outputs of a CPC network with k-means. Then, we train an additional autoregressive model to classify the previously obtained pseudo-labels in a supervised manner. Phoneme discriminative representation is achieved by executing the second-round clustering with the outputs of the final layer of the autoregressive model. We show that replacing a Transformer layer with a Conformer layer leads to a further gain in a lexical metric. Experimental results show that a relative improvement of 35% in a phonetic metric, 1.5% in the lexical metric, and 2.3% in a syntactic metric are achieved compared to a baseline method of CPC-small which is trained on LibriSpeech 460h data. We achieve top results in this challenge with the syntactic metric.