Supervector extraction for encoding speaker and phrase information with neural networks for text-dependent speaker verification

In this paper, we propose a new differentiable neural network with an alignment mechanism for text-dependent speaker verification. Unlike previous works, we do not extract the embedding of an utterance from the global average pooling of the temporal dimension. Our system replaces this reduction mechanism by a phonetic phrase alignment model to keep the temporal structure of each phrase since the phonetic information is relevant in the verification task. Moreover, we can apply a convolutional neural network as front-end, and, thanks to the alignment process being differentiable, we can train the network to produce a supervector for each utterance that will be discriminative to the speaker and the phrase simultaneously. This choice has the advantage that the supervector encodes the phrase and speaker information providing good performance in text-dependent speaker verification tasks. The verification process is performed using a basic similarity metric. The new model using alignment to produce supervectors was evaluated on the RSR2015-Part I database, providing competitive results compared to similar size networks that make use of the global average pooling to extract embeddings. Furthermore, we also evaluated this proposal on the RSR2015-Part II. To our knowledge, this system achieves the best published results obtained on this second part

Class token and knowledge distillation for multi-head self-attention speaker verification systems

This paper explores three novel approaches to improve the performance of speaker verification (SV) systems based on deep neural networks (DNN) using Multi-head Self-Attention (MSA) mechanisms and memory layers. Firstly, we propose the use of a learnable vector called Class token to replace the average global pooling mechanism to extract the embeddings. Unlike global average pooling, our proposal takes into account the temporal structure of the input what is relevant for the text-dependent SV task. The class token is concatenated to the input before the first MSA layer, and its state at the output is used to predict the classes. To gain additional robustness, we introduce two approaches. First, we have developed a new sampling estimation of the class token. In this approach, the class token is obtained by sampling from a list of several trainable vectors. This strategy introduces uncertainty that helps to generalize better compared to a single initialization as it is shown in the experiments. Second, we have added a distilled representation token for training a teacher-student pair of networks using the Knowledge Distillation (KD) philosophy, which is combined with the class token. This distillation token is trained to mimic the predictions from the teacher network, while the class token replicates the true label. All the strategies have been tested on the RSR2015-Part II and DeepMine-Part 1 databases for text-dependent SV, providing competitive results compared to the same architecture using the average pooling mechanism to extract average embeddings

The effects of child language development on the performance of automatic speech recognition

In comparison to adults’, children’s ASR appears to be more challenging and yields inferior results. It has been suggested that for this issue to be addressed, linguistic understanding of children’s speech development needs to be employed to either provide a solution or an explanation. The present work aims to explore the influence of phonological effects associated with language acquisition (PEALA) in children’s ASR and investigate whether they can be detected in systematic patterns of ASR phone confusion errors or they can be evidenced in systematic patterns of acoustic feature structure. Findings from speech development research are used as the framework upon which a set of predictable error patterns is defined and guides the analysis of the experimental results reported. Several ASR experiments are conducted involving both children’s and adults’ speech. ASR phone confusion matrices are extracted and analysed according to a statistical significance test, proposed for the purposes of this work. A mathematical model is introduced to interpret the emerging results. Additionally, bottleneck features and i-vectors representing the acoustic features in one of the systems developed, are extracted and visualised using linear discriminant analysis (LDA). A qualitative analysis is conducted with reference to patterns that can be predicted through PEALA

An Ordinal Approach to Affective Computing

Both depression prediction and emotion recognition systems are often based on ordinal ground truth due to subjectively annotated datasets. Yet, both have so far been posed as classification or regression problems. These naive approaches have fundamental issues because they are not focused on ordering, unlike ordinal regression, which is the most appropriate for truly ordinal ground truth. Ordinal regression to date offers comparatively fewer, more limited methods when compared with other branches in machine learning, and its usage has been limited to specific research domains. Accordingly, this thesis presents investigations into ordinal approaches for affective computing by describing a consistent framework to understand all ordinal system designs, proposing ordinal systems for large datasets, and introducing tools and principles to select suitable system designs and evaluation methods. First, three learning approaches are compared using the support vector framework to establish the empirical advantages of ordinal regression, which is lacking from the current literature. Results on depression and emotion corpora indicate that ordinal regression with proper tuning can improve existing depression and emotion systems. Ordinal logistic regression (OLR), which is an extension of logistic regression for ordinal scales, contributes to a number of model structures, from which the best structure must be chosen. Exploiting the newly proposed computationally efficient greedy algorithm for model structure selection (GREP), OLR outperformed or was comparable with state-of-the-art depression systems on two benchmark depression speech datasets. Deep learning has dominated many affective computing fields, and hence ordinal deep learning is an attractive prospect. However, it is under-studied even in the machine learning literature, which motivates an in-depth analysis of appropriate network architectures and loss functions. One of the significant outcomes of this analysis is the introduction of RankCNet, a novel ordinal network which utilises a surrogate loss function of rank correlation. Not only the modelling algorithm but the choice of evaluation measure depends on the nature of the ground truth. Rank correlation measures, which are sensitive to ordering, are more apt for ordinal problems than common classification or regression measures that ignore ordering information. Although rank-based evaluation for ordinal problems is not new, so far in affective computing, ordinality of the ground truth has been widely ignored during evaluation. Hence, a systematic analysis in the affective computing context is presented, to provide clarity and encourage careful choice of evaluation measures. Another contribution is a neural network framework with a novel multi-term loss function to assess the ordinality of ordinally-annotated datasets, which can guide the selection of suitable learning and evaluation methods. Experiments on multiple synthetic and affective speech datasets reveal that the proposed system can offer reliable and meaningful predictions about the ordinality of a given dataset. Overall, the novel contributions and findings presented in this thesis not only improve prediction accuracy but also encourage future research towards ordinal affective computing: a different paradigm, but often the most appropriate

Voice biometric system security: Design and analysis of countermeasures for replay attacks.

PhD ThesisVoice biometric systems use automatic speaker veri cation (ASV) technology for user authentication. Even if it is among the most convenient means of biometric authentication, the robustness and security of ASV in the face of spoo ng attacks (or presentation attacks) is of growing concern and is now well acknowledged by the research community. A spoo ng attack involves illegitimate access to personal data of a targeted user. Replay is among the simplest attacks to mount | yet di cult to detect reliably and is the focus of this thesis. This research focuses on the analysis and design of existing and novel countermeasures for replay attack detection in ASV, organised in two major parts. The rst part of the thesis investigates existing methods for spoo ng detection from several perspectives. I rst study the generalisability of hand-crafted features for replay detection that show promising results on synthetic speech detection. I nd, however, that it is di cult to achieve similar levels of performance due to the acoustically di erent problem under investigation. In addition, I show how class-dependent cues in a benchmark dataset (ASVspoof 2017) can lead to the manipulation of class predictions. I then analyse the performance of several countermeasure models under varied replay attack conditions. I nd that it is di cult to account for the e ects of various factors in a replay attack: acoustic environment, playback device and recording device, and their interactions. Subsequently, I developed and studied a convolutional neural network (CNN) model that demonstrates comparable performance to the one that ranked rst in the ASVspoof 2017 challenge. Here, the experiment analyses what the CNN has learned for replay detection using a method from interpretable machine learning. The ndings suggest that the model highly attends at the rst few milliseconds of test recordings in order to make predictions. Then, I perform an in-depth analysis of a benchmark dataset (ASVspoof 2017) for spoo ng detection and demonstrate that any machine learning countermeasure model can still exploit the artefacts I identi ed in this dataset. The second part of the thesis studies the design of countermeasures for ASV, focusing on model robustness and avoiding dataset biases. First, I proposed an ensemble model combining shallow and deep machine learning methods for spoo ng detection, and then demonstrate its e ectiveness on the latest benchmark datasets (ASVspoof 2019). Next, I proposed the use of speech endpoint detection for reliable and robust model predictions on the ASVspoof 2017 dataset. For this, I created a publicly available collection of hand-annotations of speech endpoints for the same dataset, and new benchmark results for both frame-based and utterance-based countermeasures are also developed. I then proposed spectral subband modelling using CNNs for replay detection. My results indicate that models that learn subband-speci c information substantially outperform models trained on complete spectrograms. Finally, I proposed to use variational autoencoders | deep unsupervised generative models | as an alternative backend for spoo ng detection and demonstrate encouraging results when compared with the traditional Gaussian mixture mode