This paper presents an overview of a state-of-the-art text-independent speaker verification system. First, an introduction proposes a modular scheme of the training and test phases of a speaker verification system. Then, the most commonly speech parameteriza-tion used in speaker verification, namely, cepstral analysis, is detailed. Gaussian mixture modeling, which is the speaker modeling technique used in most systems, is then explained. A few speaker modeling alternatives, namely, neural networks and support vector machines, are mentioned. Normalization of scores is then explained, as this is a very important step to deal with real-world data. The evaluation of a speaker verification system is then detailed, and the detection error trade-off (DET) curve is explained. Several extensions of speaker verification are then enumerated, including speaker tracking and segmentation by speakers. Then, some applications of speaker verification are proposed, including on-site applications, remote applications, applications relative to structuring audio information, and games. Issues concerning the forensic area are then recalled, as we believe it is very important to inform people about the actual performance and limitations of speaker verification systems. This paper concludes by giving a few research trends in speaker verification for the next couple of years.
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New waves of consumer-centric applications, such as voice search and voice interaction with mobile devices and home entertainment systems, increasingly require automatic speech recognition (ASR) to be robust to the full range of real-world noise and other acoustic distorting conditions. Despite its practical importance, however, the inherent links between and distinctions among the myriad of methods for noise-robust ASR have yet to be carefully studied in order to advance the field further. To this end, it is critical to establish a solid, consistent, and common mathematical foundation for noise-robust ASR, which is lacking at present. This article is intended to fill this gap and to provide a thorough overview of modern noise-robust techniques for ASR developed over the past 30 years. We emphasize methods that are proven to be successful and that are likely to sustain or expand their future applicability. We distill key insights from our comprehensive overview in this field and take a fresh look at a few old problems, which nevertheless are still highly relevant today. Specifically, we have analyzed and categorized a wide range of noise-robust techniques using five different criteria: 1) feature-domain vs. model-domain processing, 2) the use of prior knowledge about the acoustic environment distortion, 3) the use of explicit environment-distortion models, 4) deterministic vs. uncertainty processing, and 5) the use of acoustic models trained jointly with the same feature enhancement or model adaptation process used in the testing stage. With this taxonomy-oriented review, we equip the reader with the insight to choose among techniques and with the awareness of the performance-complexity tradeoffs. The pros and cons of using different noise-robust ASR techniques in practical application scenarios are provided as a guide to interested practitioners. The current challenges and future research directions in this field is also carefully analyzed.
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在过去几年中,自动说话人识别(ASV)的表现攻击检测(PAD)领域取得了重大进展。这包括开发新的语音语料库,标准评估协议以及前端特征提取和后端分类器的进步。 。标准数据库和评估协议的使用首次实现了对不同PAD解决方案的有意义的基准测试。本章总结了进展,重点关注过去三年完成的研究。本文概述了两个ASVspoof挑战的结果和经验教训,这是第一个以社区为主导的基准测试工作。这表明ASV PAD仍然是一个尚未解决的问题,需要进一步关注开发广泛的PAD解决方案,这些解决方案有可能检测出多样化和以前看不见的欺骗行为。攻击。
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While biometric authentication has advanced significantly in recent years, evidence shows the technology can be susceptible to malicious spoofing attacks. The research community has responded with dedicated countermeasures which aim to detect and deflect such attacks. Even if the literature shows that they can be effective, the problem is far from being solved; biometric systems remain vulnerable to spoofing. Despite a growing momentum to develop spoofing countermeasures for automatic speaker verification, now that the technology has matured sufficiently to support mass deployment in an array of diverse applications, greater effort will be needed in the future to ensure adequate protection against spoofing. This article provides a survey of past work and identifies priority research directions for the future. We summarise previous studies involving impersonation, replay, speech synthesis and voice conversion spoofing attacks and more recent efforts to develop dedicated countermeasures. The survey shows that future research should address the lack of standard datasets and the over-fitting of existing countermeasures to specific, known spoofing attacks.
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A tutorial on the design and development of automatic speaker-recognition systems is presented. Automatic speaker recognition is the use of a machine to recognize a person from a spoken phrase. These systems can operate in two modes: to identify a particular person or to verify a person's claimed identity. Speech processing and the basic components of automatic speaker-recognition systems are shown and design tradeoffs are discussed. Then, a new automatic speaker-recognition system is given. This recognizer performs with 98.9% correct identification. Last, the performances of various systems are compared.
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This paper introduces recent advances in speaker recognition technology. The first part discusses general topics and issues. The second part is devoted to a discussion of more specific topics of recent interest that have led to interesting new approaches and techniques. They include VQ-and ergodic-HMM-based text-independent recognition methods, a text-prompted recognition method, parameter/ distance normalization and model adaptation techniques, and methods of updating models and a priori thresholds in speaker verification. Although many recent advances and successes have been achieved in speaker recognition, there are still many problems for which good solutions remain to be found. The last part of this paper describes 16 open questions about speaker recognition. The paper concludes with a short discussion assessing the current status and future possibilities.
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In this paper we describe the major elements of MIT Lincoln Labo-ratory's Gaussian mixture model (GMM)-based speaker verification system used successfully in several NIST Speaker Recognition Evaluations (SREs). The system is built around the likelihood ratio test for verification, using simple but effective GMMs for likelihood functions, a universal background model (UBM) for alternative speaker representation, and a form of Bayesian adaptation to derive speaker models from the UBM. The development and use of a handset detector and score normalization to greatly improve verification performance is also described and discussed. Finally, representative performance benchmarks and system behavior experiments on NIST SRE corpora are presented.
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Voice transformation (VT) aims to change one or more aspects of a speech signal while preserving linguistic information. A subset of VT, Voice conversion (VC) specifically aims to change a source speaker's speech in such a way that the generated output is perceived as a sentence uttered by a target speaker. Despite many years of research, VC systems still exhibit deficiencies in accurately mimicking a target speaker spectrally and prosodically, and simultaneously maintaining high speech quality. In this work we provide an overview of real-world applications, extensively study existing systems proposed in the literature, and discuss remaining challenges.
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Automatic Speech Recognition (ASR) has historically been a driving force behind many machine learning (ML) techniques, including the ubiquitously used hidden Markov model, discriminative learning, structured sequence learning, Bayesian learning, and adaptive learning. Moreover, ML can and occasionally does use ASR as a large-scale, realistic application to rigorously test the effectiveness of a given technique, and to inspire new problems arising from the inherently sequential and dynamic nature of speech. On the other hand, even though ASR is available commercially for some applications, it is largely an unsolved problem-for almost all applications, the performance of ASR is not on par with human performance. New insight from modern ML methodology shows great promise to advance the state-of-the-art in ASR technology. This overview article provides readers with an overview of modern ML techniques as utilized in the current and as relevant to future ASR research and systems. The intent is to foster further cross-pollination between the ML and ASR communities than has occurred in the past. The article is organized according to the major ML paradigms that are either popular already or have potential for making significant contributions to ASR technology. The paradigms presented and elaborated in this overview include: generative and discriminative learning; supervised, unsupervised, semi-supervised, and active learning; adaptive and multi-task learning; and Bayesian learning. These learning paradigms are motivated and discussed in the context of ASR technology and applications. We finally present and analyze recent developments of deep learning and learning with sparse representations, focusing on their direct relevance to advancing ASR technology.
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Recently, increasing attention has been directed to the study of the emotional content of speech signals, and hence, many systems have been proposed to identify the emotional content of a spoken utterance. This paper is a survey of speech emotion classification addressing three important aspects of the design of a speech emotion recognition system. The first one is the choice of suitable features for speech representation. The second issue is the design of an appropriate classification scheme and the third issue is the proper preparation of an emotional speech database for evaluating system performance. Conclusions about the performance and limitations of current speech emotion recognition systems are discussed in the last section of this survey. This section also suggests possible ways of improving speech emotion recognition systems.
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声学数据提供从生物学和通信到海洋和地球科学等领域的科学和工程见解。我们调查了机器学习(ML)的进步和变革潜力,包括声学领域的深度学习。 ML是用于自动检测和利用模式印度的广泛的统计技术家族。相对于传统的声学和信号处理,ML是数据驱动的。给定足够的训练数据,ML可以发现特征之间的复杂关系。通过大量的训练数据,ML candiscover模型描述复杂的声学现象,如人类语音和混响。声学中的ML正在迅速发展,具有令人瞩目的成果和未来的重大前景。我们首先介绍ML,然后在五个声学研究领域强调MLdevelopments:语音处理中的源定位,海洋声学中的源定位,生物声学,地震探测和日常场景中的环境声音。
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In this paper, we describe recent work at Idiap Research Institute in the domain of multilingual speech processing and provide some insights into emerging challenges for the research community. Multilingual speech processing has been a topic of ongoing interest to the research community for many years and the field is now receiving renewed interest owing to two strong driving forces. Firstly, technical advances in speech recognition and synthesis are posing new challenges and opportunities to researchers. For example, discriminative features are seeing wide application by the speech recognition community, but additional issues arise when using such features in a multilingual setting. Another example is the apparent convergence of speech recognition and speech synthesis technologies in the form of statistical para-metric methodologies. This convergence enables the investigation of new approaches to unified modelling for automatic speech recognition and text-to-speech synthesis (TTS) as well as cross-lingual speaker adaptation for TTS. The second driving force is the impetus being provided by both government and industry for technologies to help break down domestic and international language barriers, these also being barriers to the expansion of policy and commerce. Speech-to-speech and speech-to-text translation are thus emerging as key technologies at the heart of which lies multilingual speech processing.
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In this work, we compare the performance of three modern speaker verification systems and non-expert human listeners in the presence of voice mimicry. Our goal is to gain insights on how vulnerable speaker verification systems are to mimicry attack and compare it to the performance of human listeners. We study both traditional Gaussian mixture model-universal background model (GMM-UBM) and an i-vector based classifier with cosine scoring and probabilistic linear discriminant analysis (PLDA) scoring. For the studied material in Finnish language, the mimicry attack decreased lightly the equal error rate (EER) for GMM-UBM from 10.83 to 10.31, while for i-vector systems the EER increased from 6.80 to 13.76 and from 4.36 to 7.38. The performance of the human listening panel shows that imitated speech increases the difficulty of the speaker verification task. It is even more difficult to recognize a person who is intentionally concealing his or her identity. For Impersonator A, the average listener made 8 errors from 34 trials while the automatic systems had 6 errors in the same set. The average listener for Impersonator B made 7 errors from the 28 trials, while the automatic systems made 7 to 9 errors. A statistical analysis of the listener performance was also conducted. We found out a statistically significant association, with p ¼ 0:00019 and R 2 ¼ 0:59, between listener accuracy and self reported factors only when familiar voices were present in the test.
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It is suggested that algorithms capable of estimating and characterizing accent knowledge would provide valuable information in the development of more effective speech systems such as speech recognition, speaker identification, audio stream tagging in spoken document retrieval, channel monitoring, or voice conversion. Accent knowledge could be used for selection of alternative pronunciations in a lexicon, engage adaptation for acoustic mod-eling, or provide information for biasing a language model in large vocabulary speech recognition. In this paper, we propose a text-independent automatic accent classification system using phone-based models. Algorithm formulation begins with a series of experiments focused on capturing the spectral evolution information as potential accent sensitive cues. Alternative subspace representations using principal component analysis and linear discriminant analysis with projected trajectories are considered. Finally, an experimental study is performed to compare the spectral trajectory model framework to a traditional hidden Markov model recognition framework using an accent sensitive word corpus. System evaluation is performed using a corpus representing five English speaker groups with native American English, and English spoken with Mandarin Chinese, French, Thai, and Turkish accents for both male and female speakers.
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This review gives a general overview of techniques used in statistical parametric speech synthesis. One instance of these techniques, called hidden Markov model (HMM)-based speech synthesis, has recently been demonstrated to be very effective in synthesizing acceptable speech. This review also contrasts these techniques with the more conventional technique of unit-selection synthesis that has dominated speech synthesis over the last decade. The advantages and drawbacks of statistical parametric synthesis are highlighted and we identify where we expect key developments to appear in the immediate future.
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Despite surveillance systems are becoming increasingly ubiquitous in our living environment, automated surveillance, currently based on video sensory modality and machine intelligence, lacks most of the time the ro-bustness and reliability required in several real applications. To tackle this issue, audio sensory devices have been taken into account, both alone or in combination with video, giving birth, in the last decade, to a considerable amount of research. In this paper audio-based automated surveillance methods are organized into a comprehensive survey: a general taxonomy, inspired by the more widespread video surveillance field, is proposed in order to systematically describe the methods covering background subtraction, event classification, object tracking and situation analysis. For each of these tasks, all the significant works are reviewed, detailing their pros and cons and the context for which they have been proposed. Moreover, a specific section is devoted to audio features, discussing their expressiveness and their employment in the above described tasks. Differently, from other surveys on audio processing and analysis, the present one is specifically targeted to automated surveillance, highlighting the target applications of each described methods and providing the reader tables and schemes useful to retrieve the most suited algorithms for a specific requirement.
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We have made signiicant progress in automatic speech recognition ASR for well-deened applications like dictation and medium vocabulary transaction processing tasks in relatively controlled environments. However, for ASR to approach h uman levels of performance and for speech to become a truly pervasive user interface, we need novel, nontraditional approaches that have the potential of yielding dramatic ASR improvements. Visual speech is one such source for making large improvements in high noise environments with the potential of channel and task independence. It is not eeected by the acoustic environment and noise, and it possibly contains the greatest amount of complementary information to the acoustic signal. In this workshop, our goal was to advance the state-of-the-art in ASR by demonstrating the use of visual information in addition to the traditional audio for large vocabulary continuous speech recognition LVCSR. Starting with an appropriate audiovisual database, collected and provided by IBM, we demonstrated for the rst time that LVCSR performance can be improved by the use of visual information in the clean audio case. Speciically, b y conducting audio lattice rescoring experiments, we showed a 7 relative word error rate WER reduction in that condition. Furthermore, for the harder problem of speech contaminated by s p e e c h babble" noise at 10 dB SNR, we demonstrated that recognition performance can beimproved by 27 in relative WER reduction, compared to an equivalent audio-only rec-ognizer matched to the noise environment. We believe that this paves the way to seriously address the challenge of speech recognition in high noise environments and to potentially achieve human levels of performance. In this report, we detail a number of approaches and experiments conducted during the summer workshop in the areas of visual feature extraction , hidden Markov model based visual-only recognition, and audiovisual information fusion. The later was our main concentration: In the workshop, a numberof feature fusion as well as decision fusion techniques for audiovisual ASR were explored and compared.
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Recently deep learning has been successfully used in speech recognition, however it has not been carefully explored and widely accepted for speaker verification. To incorporate deep learning into speaker verification, this paper proposes novel approaches of extracting and using features from deep learning models for text-dependent speaker verification. In contrast to the traditional short-term spectral feature, such as MFCC or PLP, in this paper, outputs from hidden layer of various deep models are employed as deep features for text-dependent speaker verification. Fours types of deep models are investigated: deep Restricted Boltzmann Machines, speech-discriminant Deep Neural Network (DNN), speaker-discriminant DNN, and multi-task joint-learned DNN. Once deep features are extracted, they may be used within either the GMM-UBM framework or the identity vector (i-vector) framework. Joint linear dis-criminant analysis and probabilistic linear discriminant analysis are proposed as effective back-end classifiers for identity vector based deep features. These approaches were evaluated on the RSR2015 data corpus. Experiments showed that deep feature based methods can obtain significant performance improvements compared to the traditional baselines, no matter if they are directly applied in the GMM-UBM system or utilized as identity vectors. The EER of the best system using the proposed identity vector is 0.10%, only one fifteenth of that in the GMM-UBM baseline.
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Speaker diarization is the task of determining "who spoke when?" in an audio or video recording that contains an unknown amount of speech and also an unknown number of speakers. Initially, it was proposed as a research topic related to automatic speech recognition, where speaker diarization serves as an upstream processing step. Over recent years, however, speaker diarization has become an important key technology for many tasks, such as navigation, retrieval, or higher-level inference on audio data. Accordingly, many important improvements in accuracy and robustness have been reported in journals and conferences in the area. The application domains, from broadcast news, to lectures and meetings, vary greatly and pose different problems, such as having access to multiple microphones and multimodal information or overlapping speech. The most recent review of existing technology dates back to 2006 and focuses on the broadcast news domain. In this paper we review the current state-of-the-art, focusing on research developed since 2006 that relates predominantly to speaker diarization for conference meetings. Finally, we present an analysis of speaker diarization performance as reported through the NIST Rich Transcription evaluations on meeting data and identify important areas for future research.
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