89,985 research outputs found

    Continuous Audio-Visual Speech Recognition

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    We address the problem of robust lip tracking, visual speech feature extraction, and sensor integration for audio-visual speech recognition applications. An appearance based model of the articulators, which represents linguistically important features, is learned from example images and is used to locate, track, and recover visual speech information. We tackle the problem of joint temporal modelling of the acoustic and visual speech signals by applying Multi-Stream hidden Markov models. This approach allows the use of different temporal topologies and levels of stream integration and hence enables to model temporal dependencies more accurately. The system has been evaluated for a continuously spoken digit recognition task of 37 subjects

    Using the Multi-Stream Approach for Continuous Audio-Visual Speech Recognition: Experiments on the M2VTS Database

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    The Multi-Stream automatic speech recognition approach was investigated in this work as a framework for Audio-Visual data fusion and speech recognition. This method presents many potential advantages for such a task. It particularly allows for synchronous decoding of continuous speech while still allowing for some asynchrony of the visual and acoustic information streams. First, the Multi-Stream formalism is briefly recalled. Then, on top of the Multi-Stream motivations, experiments on the M2VTS multimodal database are presented and discussed. To our knowledge, these are the first experiments about multi-speaker continuous Audio-Visual Speech Recognition (AVSR). It is shown that the Multi-Stream approach can yield improved Audio-Visual speech recognition performance when the acoustic signal is corrupted by noise as well as for clean speech

    Using the Multi-Stream Approach for Continuous Audio-Visual Speech Recognition

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    The Multi-Stream automatic speech recognition approach was investigated in this work as a framework for Audio-Visual data fusion and speech recognition. This method presents many potential advantages for such a task. It particularly allows for synchronous decoding of continuous speech while still allowing for some asynchrony of the visual and acoustic information streams. First, the Multi-Stream formalism is briefly recalled. Then, on top of the Multi-Stream motivations, experiments on the {\sc M2VTS} multimodal database are presented and discussed. To our knowledge, these are the first experiments about multi-speaker continuous Audio-Visual Speech Recognition (AVSR). It is shown that the Multi-Stream approach can yield improved Audio-Visual speech recognition performance when the acoustic signal is corrupted by noise as well as for clean speech

    A new multi-modal database for developing speech recognition systems for an assistive technology application

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    In this paper we report on the acquisition and content of a new database intended for developing audio-visual speech recognition systems. This database supports a speaker dependent continuous speech recognition task, based on a small vocabulary, and was captured in the European Portuguese language. Along with the collected multi-modal speech materials, the respective orthographic transcription and time-alignment files are supplied. The package also includes data on stochastic language models and the generative grammar associated to the collected spoken sentences. The application addressed by this database, which consists of voice control of a basic scientific calculator, has the particularity of being designed for a person with a specific motor impairment, namely muscular dystrophy. This specificity is a remarkable characteristic, given the lack of such kind of data resources for developing assistive systems based on audio-visual speech recognition technology

    Deep audio-visual speech recognition

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    Decades of research in acoustic speech recognition have led to systems that we use in our everyday life. However, even the most advanced speech recognition systems fail in the presence of noise. The degraded performance can be compensated by introducing visual speech information. However, Visual Speech Recognition (VSR) in naturalistic conditions is very challenging, in part due to the lack of architectures and annotations. This thesis contributes towards the problem of Audio-Visual Speech Recognition (AVSR) from different aspects. Firstly, we develop AVSR models for isolated words. In contrast to previous state-of-the-art methods that consists of a two-step approach, feature extraction and recognition, we present an End-to-End (E2E) approach inside a deep neural network, and this has led to a significant improvement in audio-only, visual-only and audio-visual experiments. We further replace Bi-directional Gated Recurrent Unit (BGRU) with Temporal Convolutional Networks (TCN) to greatly simplify the training procedure. Secondly, we extend our AVSR model for continuous speech by presenting a hybrid Connectionist Temporal Classification (CTC)/Attention model, that can be trained in an end-to-end manner. We then propose the addition of prediction-based auxiliary tasks to a VSR model and highlight the importance of hyper-parameter optimisation and appropriate data augmentations. Next, we present a self-supervised framework, Learning visual speech Representations from Audio via self-supervision (LiRA). Specifically, we train a ResNet+Conformer model to predict acoustic features from unlabelled visual speech, and find that this pre-trained model can be leveraged towards word-level and sentence-level lip-reading. We also investigate the Lombard effect influence in an end-to-end AVSR system, which is the first work using end-to-end deep architectures and presents results on unseen speakers. We show that even if a relatively small amount of Lombard speech is added to the training set then the performance in a real scenario, where noisy Lombard speech is present, can be significantly improved. Lastly, we propose a detection method against adversarial examples in an AVSR system, where the strong correlation between audio and visual streams is leveraged. The synchronisation confidence score is leveraged as a proxy for audio-visual correlation and based on it, we can detect adversarial attacks. We apply recent adversarial attacks on two AVSR models and the experimental results demonstrate that the proposed approach is an effective way for detecting such attacks.Open Acces

    History-based visual mining of semi-structured audio and text

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    Accessing specific or salient parts of multimedia recordings remains a challenge as there is no obvious way of structuring and representing a mix of space-based and time-based media. A number of approaches have been proposed which usually involve translating the continuous component of the multimedia recording into a space-based representation, such as text from audio through automatic speech recognition and images from video (keyframes). In this paper, we present a novel technique which defines retrieval units in terms of a log of actions performed on space-based artefacts, and exploits timing properties and extended concurrency to construct a visual presentation of text and speech data. This technique can be easily adapted to any mix of space-based artefacts and continuous media

    Improving Speech-related Facial Action Unit Recognition by Audiovisual Information Fusion

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    In spite of great progress achieved on posed facial display and controlled image acquisition, performance of facial action unit (AU) recognition degrades significantly for spontaneous facial displays. Furthermore, recognizing AUs accompanied with speech is even more challenging since they are generally activated at a low intensity with subtle facial appearance/geometrical changes during speech, and more importantly, often introduce ambiguity in detecting other co-occurring AUs, e.g., producing non-additive appearance changes. All the current AU recognition systems utilized information extracted only from visual channel. However, sound is highly correlated with visual channel in human communications. Thus, we propose to exploit both audio and visual information for AU recognition. Specifically, a feature-level fusion method combining both audio and visual features is first introduced. Specifically, features are independently extracted from visual and audio channels. The extracted features are aligned to handle the difference in time scales and the time shift between the two signals. These temporally aligned features are integrated via feature-level fusion for AU recognition. Second, a novel approach that recognizes speech-related AUs exclusively from audio signals based on the fact that facial activities are highly correlated with voice during speech is developed. Specifically, dynamic and physiological relationships between AUs and phonemes are modeled through a continuous time Bayesian network (CTBN); then AU recognition is performed by probabilistic inference via the CTBN model. Third, a novel audiovisual fusion framework, which aims to make the best use of visual and acoustic cues in recognizing speech-related facial AUs is developed. In particular, a dynamic Bayesian network (DBN) is employed to explicitly model the semantic and dynamic physiological relationships between AUs and phonemes as well as measurement uncertainty. AU recognition is then conducted by probabilistic inference via the DBN model. To evaluate the proposed approaches, a pilot AU-coded audiovisual database was collected. Experiments on this dataset have demonstrated that the proposed frameworks yield significant improvement in recognizing speech-related AUs compared to the state-of-the-art visual-based methods. Furthermore, more impressive improvement has been achieved for those AUs, whose visual observations are impaired during speech
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