29 accepted papers
This paper investigates cascade approaches bridging speech and text foundation models (FMs) for speech translation (ST). We address the limitations of cascade systems which suffer from the propagation of speech recognition errors and the lack of access to acoustic information. We propose a ReShape A…
This paper proposes a novel speech emotion recognition (SER) method that fully leverages the architecture of Whisper, a large-scale automatic speech recognition (ASR) model. The conventional SER models using a pre-trained speech encoder may fail to capture linguistic content since their decoders are…
This paper details our speaker diarization system designed for multi-domain, multi-microphone casual conversations. The proposed diarization pipeline uses weighted prediction error (WPE)based dereverberation as a front end, and separately applies end-to-end neural diarization with vector clustering…
The quadratic memory complexity of self-attention has generally restricted Transformer-based models to utterance-based speech processing, preventing models from leveraging long-form contexts. A common solution has been to formulate long-form speech processing into a streaming problem, only using lim…
We propose a new shallow fusion (SF) method to exploit an external backward language model (BLM) for end-to-end automatic speech recognition (ASR). The BLM has complementary characteristics with a forward language model (FLM), and the effectiveness of their combination has been confirmed by rescorin…
End-to-end speech summarization (E2E SSum) is a technique to directly generate summary sentences from speech. Compared with the cascade approach, which combines automatic speech recognition (ASR) and text summarization models, the E2E approach is more promising because it mitigates ASR errors, incor…
Speech summarization requires processing several minute-long speech sequences to allow exploiting the whole context of a spoken document. A conventional approach is a cascade of automatic speech recognition (ASR) and text summarization (TS). However, the cascade systems are sensitive to ASR errors.…
Spoken language processing (SLP) systems such as speech summarization and translation can be achieved by cascade models. It combines an automatic speech recognition (ASR) frontend and a natural language processing (NLP) backend including machine translation (MT) or text summarization (TS). With this…
We investigate the effectiveness of using a large ensemble of advanced neural language models (NLMs) for lattice rescoring on automatic speech recognition (ASR) hypotheses. Previous studies have reported the effectiveness of combining a small number of NLMs. In contrast, in this study, we combine up…
Estimating a speaker’s age from their speech is more challenging than age estimation from their face because of insufficiently available public corpora. To tackle this problem, we construct a new audio-visual age corpus named AgeVoxCeleb by annotating age labels to VoxCeleb2 videos. AgeVoxCeleb is t…
Confidence estimation, in which we estimate the reliability of each recognized token (e.g., word, sub-word, and character) in automatic speech recognition (ASR) hypotheses and detect incorrectly recognized tokens, is an important function for developing ASR applications. In this study, we perform co…
This paper investigates a phoneme-invariant speaker embedding approach for speaker recognition on extremely short utterances. Intuitively, phonemes are nuisance information for text-independent speaker recognition task since the contents of the speech are usually mismatched between enrolling and tes…
This paper proposes a general post-processing method for improving speaker-attribute estimation. Estimating speaker-specific attributes such as age and gender is an important task with a wide range of applications. While the recent proposed deep neural network-based end-to-end approach achieves high…
An important task for language models is the adaptation of general-domain models to specific target domains. For neural network-based language models, feature-based domain adaptation has been a popular method in previous research. Conventional methods use an adaptation feature providing context info…
The development of deep learning techniques has triggered the active investigation of neural network-based speech enhancement approaches. In particular, single-channel blind (uninformed) speech separation and speaker-aware (informed) speech extraction have received increased interest. Blind speech s…
We propose an integer linear programming (ILP)-based compressive speech summarization method that maximizes the coverage of content words in a resultant summary. It is an unsupervised method and, under the designed constraints, it performs a single-step globally optimal summarization of a given long…
We introduce speech and text autoencoders that share encoders and decoders with an automatic speech recognition (ASR) model to improve ASR performance with large speech only and text only training datasets. To build the speech and text autoencoders, we leverage state-of-the-art ASR and text-to-speec…
Training recurrent neural network language models (RNNLMs) requires a large amount of data, which is difficult to collect for specific domains such as multiparty conversations. Data augmentation using external resources and model adaptation, which adjusts a model trained on a large amount of data to…
This paper proposes a new model for accurately rescoring (reranking) N-best speech recognition hypothesis lists. The model is based on state-of-the-art neural networks (NNs) and provides the minimum necessary functionality to perform N-best rescoring, i.e. one-on-one hypothesis comparison on a given…
The standard evaluation metric of automatic speech recognition (ASR) is the word error rate (WER), which measures the dissimilarity between recognized word sequences and their ground truth. Many training algorithms designed to reduce sequence-level errors such as WER have been proposed for hidden Ma…
This paper addresses the problem of single channel speech recognition of a target speaker in a mixture of speech signals. We propose to exploit auxiliary speaker information provided by an adaptation utterance from the target speaker to extract and recognize only that speaker. Using such auxiliary i…
Adapting acoustic models to speakers have shown to greatly improve performance for many tasks. Among the adaptation approaches, exploiting auxiliary features characterizing speakers or environments has received great attention because they allow rapid adaptation, i.e. adaptation with limited amount…
We propose a novel framework designed to extend conventional deep neural network (DNN)-based feature enhancement approaches. In general, the conventional DNN-based feature enhancement framework aims to map input noisy observation to clean speech or a binary/ soft mask in a deterministic way, assumin…
The use of auxiliary features is an effective way to improve the performance of deep neural network (DNN)-based acoustic models. Most approaches use auxiliary features that represent the speaker or the environment. These auxiliary features are usually computed independently of the acoustic model. Th…
We propose a new concept for adapting CNN-based acoustic models using spatial diffuseness features as auxiliary information about the acoustic environment: the spatial diffuseness features are simultaneously employed as acoustic-model input features and to estimate environmental cues for context ada…
Deep neural network (DNN) based acoustic models have greatly improved the performance of automatic speech recognition (ASR) for various tasks. Further performance improvements have been reported when making DNNs aware of the acoustic context (e.g. speaker or environment) for example by adding auxili…
This paper addresses a minimum variance distortionless response (MVDR) beamforming based speech enhancement approach for meeting speech recognition. In a meeting situation, speaker overlaps and noise signals are not negligible. To handle these issues, we employ MVDR beamforming, where accurate estim…
Recurrent neural networks (RNNs) have recently been applied as the classifiers for sequential labeling problems. In this paper, deep bidirectional RNNs (DBRNNs) are applied for the first time to error detection in automatic speech recognition (ASR), which is a sequential labeling problem. We investi…
This paper presents a novel double-layer neighborhood graph index for acceleration of similarity search that accomplishes fast querybyexample spoken term detection (STD). When a query segment is given, our proposed STD method finds similar segments to the query from an utterance data set by efficien…