Robust text independent closed set speaker identification systems and their evaluation

PhD ThesisThis thesis focuses upon text independent closed set speaker identi cation. The contributions relate to evaluation studies in the presence of various types of noise and handset e ects. Extensive evaluations are performed on four databases. The rst contribution is in the context of the use of the Gaussian Mixture Model-Universal Background Model (GMM-UBM) with original speech recordings from only the TIMIT database. Four main simulations for Speaker Identi cation Accuracy (SIA) are presented including di erent fusion strategies: Late fusion (score based), early fusion (feature based) and early-late fusion (combination of feature and score based), late fusion using concatenated static and dynamic features (features with temporal derivatives such as rst order derivative delta and second order derivative delta-delta features, namely acceleration features), and nally fusion of statistically independent normalized scores. The second contribution is again based on the GMM-UBM approach. Comprehensive evaluations of the e ect of Additive White Gaussian Noise (AWGN), and Non-Stationary Noise (NSN) (with and without a G.712 type handset) upon identi cation performance are undertaken. In particular, three NSN types with varying Signal to Noise Ratios (SNRs) were tested corresponding to: street tra c, a bus interior and a crowded talking environment. The performance evaluation also considered the e ect of late fusion techniques based on score fusion, namely mean, maximum, and linear weighted sum fusion. The databases employed were: TIMIT, SITW, and NIST 2008; and 120 speakers were selected from each database to yield 3,600 speech utterances. The third contribution is based on the use of the I-vector, four combinations of I-vectors with 100 and 200 dimensions were employed. Then, various fusion techniques using maximum, mean, weighted sum and cumulative fusion with the same I-vector dimension were used to improve the SIA. Similarly, both interleaving and concatenated I-vector fusion were exploited to produce 200 and 400 I-vector dimensions. The system was evaluated with four di erent databases using 120 speakers from each database. TIMIT, SITW and NIST 2008 databases were evaluated for various types of NSN namely, street-tra c NSN, bus-interior NSN and crowd talking NSN; and the G.712 type handset at 16 kHz was also applied. As recommendations from the study in terms of the GMM-UBM approach, mean fusion is found to yield overall best performance in terms of the SIA with noisy speech, whereas linear weighted sum fusion is overall best for original database recordings. However, in the I-vector approach the best SIA was obtained from the weighted sum and the concatenated fusion.Ministry of Higher Education and Scienti c Research (MoHESR), and the Iraqi Cultural Attach e, Al-Mustansiriya University, Al-Mustansiriya University College of Engineering in Iraq for supporting my PhD scholarship

Design Guidelines for Inclusive Speaker Verification Evaluation Datasets

Speaker verification (SV) provides billions of voice-enabled devices with access control, and ensures the security of voice-driven technologies. As a type of biometrics, it is necessary that SV is unbiased, with consistent and reliable performance across speakers irrespective of their demographic, social and economic attributes. Current SV evaluation practices are insufficient for evaluating bias: they are over-simplified and aggregate users, not representative of real-life usage scenarios, and consequences of errors are not accounted for. This paper proposes design guidelines for constructing SV evaluation datasets that address these short-comings. We propose a schema for grading the difficulty of utterance pairs, and present an algorithm for generating inclusive SV datasets. We empirically validate our proposed method in a set of experiments on the VoxCeleb1 dataset. Our results confirm that the count of utterance pairs/speaker, and the difficulty grading of utterance pairs have a significant effect on evaluation performance and variability. Our work contributes to the development of SV evaluation practices that are inclusive and fair

UNSUPERVISED DOMAIN ADAPTATION FOR SPEAKER VERIFICATION IN THE WILD

Performance of automatic speaker verification (ASV) systems is very sensitive to mismatch between training (source) and testing (target) domains. The best way to address domain mismatch is to perform matched condition training – gather sufficient labeled samples from the target domain and use them in training. However, in many cases this is too expensive or impractical. Usually, gaining access to unlabeled target domain data, e.g., from open source online media, and labeled data from other domains is more feasible. This work focuses on making ASV systems robust to uncontrolled (‘wild’) conditions, with the help of some unlabeled data acquired from such conditions. Given acoustic features from both domains, we propose learning a mapping function – a deep convolutional neural network (CNN) with an encoder-decoder architecture – between features of both the domains. We explore training the network in two different scenarios: training on paired speech samples from both domains and training on unpaired data. In the former case, where the paired data is usually obtained via simulation, the CNN is treated as a nonii ABSTRACT linear regression function and is trained to minimize L2 loss between original and predicted features from target domain. We provide empirical evidence that this approach introduces distortions that affect verification performance. To address this, we explore training the CNN using adversarial loss (along with L2), which makes the predicted features indistinguishable from the original ones, and thus, improve verification performance. The above framework using simulated paired data, though effective, cannot be used to train the network on unpaired data obtained by independently sampling speech from both domains. In this case, we first train a CNN using adversarial loss to map features from target to source. We, then, map the predicted features back to the target domain using an auxiliary network, and minimize a cycle-consistency loss between the original and reconstructed target features. Our unsupervised adaptation approach complements its supervised counterpart, where adaptation is done using labeled data from both domains. We focus on three domain mismatch scenarios: (1) sampling frequency mismatch between the domains, (2) channel mismatch, and (3) robustness to far-field and noisy speech acquired from wild conditions

MAVD: The First Open Large-Scale Mandarin Audio-Visual Dataset with Depth Information

Audio-visual speech recognition (AVSR) gains increasing attention from researchers as an important part of human-computer interaction. However, the existing available Mandarin audio-visual datasets are limited and lack the depth information. To address this issue, this work establishes the MAVD, a new large-scale Mandarin multimodal corpus comprising 12,484 utterances spoken by 64 native Chinese speakers. To ensure the dataset covers diverse real-world scenarios, a pipeline for cleaning and filtering the raw text material has been developed to create a well-balanced reading material. In particular, the latest data acquisition device of Microsoft, Azure Kinect is used to capture depth information in addition to the traditional audio signals and RGB images during data acquisition. We also provide a baseline experiment, which could be used to evaluate the effectiveness of the dataset. The dataset and code will be released at https://github.com/SpringHuo/MAVD

X-VECTORS: ROBUST NEURAL EMBEDDINGS FOR SPEAKER RECOGNITION

Speaker recognition is the task of identifying speakers based on their speech signal. Typically, this involves comparing speech from a known speaker, with recordings from unknown speakers, and making same-or-different speaker decisions. If the lexical contents of the recordings are fixed to some phrase, the task is considered text-dependent, otherwise it is text-independent. This dissertation is primarily concerned with this second, less constrained problem. Since speech data lives in a complex, high-dimensional space, it is difficult to directly compare speakers. Comparisons are facilitated by embeddings: mappings from complex input patterns to low-dimensional Euclidean spaces where notions of distance or similarity are defined in natural ways. For almost ten years, systems based on i-vectors--a type of embedding extracted from a traditional generative model--have been the dominant paradigm in this field. However, in other areas of applied machine learning, such as text or vision, embeddings extracted from discriminatively trained neural networks are the state-of-the-art. Recently, this line of research has become very active in speaker recognition as well. Neural networks are a natural choice for this purpose, as they are capable of learning extremely complex mappings, and when training data resources are abundant, tend to outperform traditional methods. In this dissertation, we develop a next-generation neural embedding--denoted by x-vector--for speaker recognition. These neural embeddings are demonstrated to substantially improve upon the state-of-the-art on a number of benchmark datasets

Face mask recognition from audio: the MASC database and an overview on the mask challenge

The sudden outbreak of COVID-19 has resulted in tough challenges for the field of biometrics due to its spread via physical contact, and the regulations of wearing face masks. Given these constraints, voice biometrics can offer a suitable contact-less biometric solution; they can benefit from models that classify whether a speaker is wearing a mask or not. This article reviews the Mask Sub-Challenge (MSC) of the INTERSPEECH 2020 COMputational PARalinguistics challengE (ComParE), which focused on the following classification task: Given an audio chunk of a speaker, classify whether the speaker is wearing a mask or not. First, we report the collection of the Mask Augsburg Speech Corpus (MASC) and the baseline approaches used to solve the problem, achieving a performance of [Formula: see text] Unweighted Average Recall (UAR). We then summarise the methodologies explored in the submitted and accepted papers that mainly used two common patterns: (i) phonetic-based audio features, or (ii) spectrogram representations of audio combined with Convolutional Neural Networks (CNNs) typically used in image processing. Most approaches enhance their models by adapting ensembles of different models and attempting to increase the size of the training data using various techniques. We review and discuss the results of the participants of this sub-challenge, where the winner scored a UAR of [Formula: see text]. Moreover, we present the results of fusing the approaches, leading to a UAR of [Formula: see text]. Finally, we present a smartphone app that can be used as a proof of concept demonstration to detect in real-time whether users are wearing a face mask; we also benchmark the run-time of the best models