Language Identification Using Visual Features

Automatic visual language identification (VLID) is the technology of using information derived from the visual appearance and movement of the speech articulators to iden- tify the language being spoken, without the use of any audio information. This technique for language identification (LID) is useful in situations in which conventional audio processing is ineffective (very noisy environments), or impossible (no audio signal is available). Research in this field is also beneficial in the related field of automatic lip-reading. This paper introduces several methods for visual language identification (VLID). They are based upon audio LID techniques, which exploit language phonology and phonotactics to discriminate languages. We show that VLID is possible in a speaker-dependent mode by discrimi- nating different languages spoken by an individual, and we then extend the technique to speaker-independent operation, taking pains to ensure that discrimination is not due to artefacts, either visual (e.g. skin-tone) or audio (e.g. rate of speaking). Although the low accuracy of visual speech recognition currently limits the performance of VLID, we can obtain an error-rate of < 10% in discriminating between Arabic and English on 19 speakers and using about 30s of visual speech

Visual speech encoding based on facial landmark registration

Visual Speech Recognition (VSR) related studies largely ignore the use of state of the art approaches in facial landmark localization, and are also deficit of robust visual features and its temporal encoding. In this work, we propose a visual speech temporal encoding by integrating state of the art fast and accurate facial landmark detection based on ensemble of regression trees learned using gradient boosting. The main contribution of this work is in proposing a fast and simple encoding of visual speech features derived from vertically symmetric point pairs (VeSPP) of facial landmarks corresponding to lip regions, and demonstrating their usefulness in temporal sequence comparisons using Dynamic Time Warping. VSR can be either speaker dependent (SD) or speaker independent (SI), and each of them poses different kind of challenges. In this work, we consider the SD scenario, and obtain 82.65% recognition accuracy on OuluVS database. Unlike recent research in VSR which makes use of auxiliary information such as audio, depth and color channels, our approach does not impose such constraints

Discriminatively trained features using fMPE for multi-stream audio-visual speech recognition

Abstract fMPE is a recently introduced discriminative training technique that uses the Minimum Phone Error (MPE) discriminative criterion to train a feature-level transformation. In this paper we investigate fMPE trained audio/visual features for multistream HMM-based audio-visual speech recognition. A flexible, layer-based implementation of fMPE allows us to combine the the visual information with the audio stream using the discriminative traning process, and dispense with the multiple stream approach. Experiments are reported on the IBM infrared headset audio-visual database. On average of 20-speaker 1 hour speaker independent test data, the fMPE trained acoustic features achieve 33% relative gain. Adding video layers on top of audio layers gives additional 10% gain over fMPE trained features from the audio stream alone. The fMPE trained visual features achieve 14% relative gain, while the decision fusion of audio/visual streams with fMPE trained features achieves 29% relative gain. However, fMPE trained models do not improve over the original models on the mismatched noisy test data

Multimodal Sensing and Data Processing for Speaker and Emotion Recognition using Deep Learning Models with Audio, Video and Biomedical Sensors

The focus of the thesis is on Deep Learning methods and their applications on multimodal data, with a potential to explore the associations between modalities and replace missing and corrupt ones if necessary. We have chosen two important real-world applications that need to deal with multimodal data: 1) Speaker recognition and identification; 2) Facial expression recognition and emotion detection. The first part of our work assesses the effectiveness of speech-related sensory data modalities and their combinations in speaker recognition using deep learning models. First, the role of electromyography (EMG) is highlighted as a unique biometric sensor in improving audio-visual speaker recognition or as a substitute in noisy or poorly-lit environments. Secondly, the effectiveness of deep learning is empirically confirmed through its higher robustness to all types of features in comparison to a number of commonly used baseline classifiers. Not only do deep models outperform the baseline methods, their power increases when they integrate multiple modalities, as different modalities contain information on different aspects of the data, especially between EMG and audio. Interestingly, our deep learning approach is word-independent. Plus, the EMG, audio, and visual parts of the samples from each speaker do not need to match. This increases the flexibility of our method in using multimodal data, particularly if one or more modalities are missing. With a dataset of 23 individuals speaking 22 words five times, we show that EMG can replace the audio/visual modalities, and when combined, significantly improve the accuracy of speaker recognition. The second part describes a study on automated emotion recognition using four different modalities – audio, video, electromyography (EMG), and electroencephalography (EEG). We collected a dataset by recording the 4 modalities as 12 human subjects expressed six different emotions or maintained a neutral expression. Three different aspects of emotion recognition were investigated: model selection, feature selection, and data selection. Both generative models (DBNs) and discriminative models (LSTMs) were applied to the four modalities, and from these analyses we conclude that LSTM is better for audio and video together with their corresponding sophisticated feature extractors (MFCC and CNN), whereas DBN is better for both EMG and EEG. By examining these signals at different stages (pre-speech, during-speech, and post-speech) of the current and following trials, we have found that the most effective stages for emotion recognition from EEG occur after the emotion has been expressed, suggesting that the neural signals conveying an emotion are long-lasting

Tandem approach for information fusion in audio visual speech recognition

Speech is the most frequently preferred medium for humans to interact with their environment making it an ideal instrument for human-computer interfaces. However, for the speech recognition systems to be more prevalent in real life applications, high recognition accuracy together with speaker independency and robustness to hostile conditions is necessary. One of the main preoccupation for speech recognition systems is acoustic noise. Audio Visual Speech Recognition systems intend to overcome the noise problem utilizing visual speech information generally extracted from the face or in particular the lip region. Visual speech information is known to be a complementary source for speech perception and is not impacted by acoustic noise. This advantage brings in two additional issues into the task which are visual feature extraction and information fusion. There is extensive research on both issues but an admissable level of success has not been reached yet. This work concentrates on the issue of information fusion and proposes a novel methodology. The aim of the proposed technique is to deploy a preliminary decision stage at frame level as an initial stage and feed the Hidden Markov Model with the output posterior probabilities as in tandem HMM approach. First, classification is performed for each modality separately. Sequentially, the individual classifiers of each modality are combined to obtain posterior probability vectors corresponding to each speech frame. The purpose of using a preliminary stage is to integrate acoustic and visual data for maximum class separability. Hidden Markov Models are employed as the second stage of modelling because of their ability to handle temporal evolutions of data. The proposed approach is investigated in a speaker independent scenario for digit recognition with the existence of diverse levels of car noise. The method is compared with a principal information fusion framework in audio visual speech recognition which is Multiple Stream Hidden Markov Models (MSHMM). The results on M2VTS database show that the novel method achieves resembling performance with less processing time as compared to MSHMM