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282
Language recognition via i-vectors and dimensionality reduction
- in Interspeech
, 2011
"... In this paper, a new language identification system is presented based on the total variability approach previously developed in the field of speaker identification. Various techniques are em-ployed to extract the most salient features in the lower dimen-sional i-vector space and the system develope ..."
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Cited by 35 (4 self)
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In this paper, a new language identification system is presented based on the total variability approach previously developed in the field of speaker identification. Various techniques are em-ployed to extract the most salient features in the lower dimen-sional i-vector space and the system developed results in excel-lent performance on the 2009 LRE evaluation set without the need for any post-processing or backend techniques. Additional performance gains are observed when the system is combined with other acoustic systems.
Cosine Similarity Scoring without Score Normalization Techniques
"... In recent work [1], a simplified and highly effective approach to speaker recognition based on the cosine similarity between lowdimensional vectors, termed ivectors, defined in a total variability space was introduced. The total variability space representation is motivated by the popular Joint Fact ..."
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Cited by 34 (2 self)
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In recent work [1], a simplified and highly effective approach to speaker recognition based on the cosine similarity between lowdimensional vectors, termed ivectors, defined in a total variability space was introduced. The total variability space representation is motivated by the popular Joint Factor Analysis (JFA) approach, but does not require the complication of estimating separate speaker and channel spaces and has been shown to be less dependent on score normalization procedures, such as z-norm and t-norm. In this paper, we introduce a modification to the cosine similarity that does not require explicit score normalization, relying instead on simple mean and covariance statistics from a collection of impostor speaker ivectors. By avoiding the complication of z- and t-norm, the new approach further allows for application of a new unsupervised speaker adaptation technique to models defined in the ivector space. Experiments are conducted on the core condition of the NIST 2008 corpora, where, with adaptation, the new approach produces an equal error rate (EER) of 4.8 % and min decision cost function (MinDCF) of 2.3 % on all female speaker trials. 1.
Leeuwen, “Source-normalised-and-weighted LDA for robust speaker recognition using i-vectors
- in accepted into IEEE Int. Conf. on Acoustics, Speech and Signal Processing
, 2011
"... The recently developed i-vector framework for speaker recognition has set a new performance standard in the research field. An i-vector is a compact representation of a speaker utterance extracted from a low-dimensional total variability subspace. Prior to classification using a cosine kernel, i-vec ..."
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Cited by 15 (5 self)
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The recently developed i-vector framework for speaker recognition has set a new performance standard in the research field. An i-vector is a compact representation of a speaker utterance extracted from a low-dimensional total variability subspace. Prior to classification using a cosine kernel, i-vectors are projected into an LDA space in order to reduce inter-session variability and enhance speaker discrimination. The accurate estimation of this LDA space from a training dataset is crucial to classification performance. A typical training dataset, however, does not consist of utterances acquired from all sources of interest (ie., telephone, microphone and interview speech sources) for each speaker. This has the effect of introducing source-related variation in the between-speaker covariance matrix and results in an incomplete representation of the within-speaker scatter matrix used for LDA. Proposed is a novel source-normalised-and-weighted LDA algorithm developed to improve the robustness of i-vector-based speaker recognition under both mis-matched evaluation conditions and conditions for which insufficient speech resources are available for adequate system development. Evaluated on the recent NIST 2008 and 2010 Speaker Recognition Evaluations (SRE), the proposed technique demonstrated improvements of up to 31 % in minimum DCF and EER under mis-matched and sparsely-resourced conditions. Index Terms — speaker recognition, linear discriminant analysis, i-vector, total variability, source variability 1.
Low-Variance Multitaper MFCC Features: a Case Study in Robust Speaker Verification
, 2012
"... In speech and audio applications, short-term signal spectrum is often represented using mel-frequency cepstral coefficients (MFCCs) computed from a windowed discrete Fourier transform (DFT). Windowing reduces spectral leakage but variance of the spectrum estimate remains high. An elegant extension ..."
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Cited by 14 (3 self)
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In speech and audio applications, short-term signal spectrum is often represented using mel-frequency cepstral coefficients (MFCCs) computed from a windowed discrete Fourier transform (DFT). Windowing reduces spectral leakage but variance of the spectrum estimate remains high. An elegant extension to windowed DFT is the so-called multitaper method which uses multiple time-domain windows (tapers) with frequencydomain averaging. Multitapers have received little attention in speech processing even though they produce low-variance features. In this paper, we propose the multitaper method for MFCC extraction with a practical focus. We provide, firstly, detailed statistical analysis of MFCC bias and variance using autoregressive process simulations on the TIMIT corpus. For speaker verification experiments on the NIST 2002 and 2008 SRE corpora, we consider three Gaussian mixture model based classifiers with universal background model (GMM-UBM), support vector machine (GMM-SVM) and joint factor analysis (GMM-JFA). Multitapers improve MinDCF over the baseline windowed DFT by relative 20.4 % (GMM-SVM) and 13.7 % (GMM-JFA) on the interview-interview condition in NIST 2008. The GMM-JFA system further reduces MinDCF by 18.7 % on the telephone data. With these improvements and generally noncritical parameter selection, multitaper MFCCs are a viable candidate for replacing the conventional MFCCs.
I4U submission to NIST SRE 2012: a large-scale collaborative effort for noise-robust speaker verification
- in InterSpeech
, 2013
"... I4U is a joint entry of nine research Institutes and Universities across 4 continents to NIST SRE 2012. It started with a brief discussion during the Odyssey 2012 workshop in Singapore. An online discussion group was soon set up, providing a dis-cussion platform for different issues surrounding NIST ..."
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Cited by 13 (5 self)
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I4U is a joint entry of nine research Institutes and Universities across 4 continents to NIST SRE 2012. It started with a brief discussion during the Odyssey 2012 workshop in Singapore. An online discussion group was soon set up, providing a dis-cussion platform for different issues surrounding NIST SRE’12. Noisy test segments, uneven multi-session training, variable en-rollment duration, and the issue of open-set identification were actively discussed leading to various solutions integrated to the I4U submission. The joint submission and several of its 17 sub-systems were among top-performing systems. We summarize the lessons learnt from this large-scale effort.
A study on spoofing attack in state-of-the-art speaker verification: the telephone speech case
- in APSIPA ASC
, 2012
"... Abstract—Voice conversion technique, which modifies one speaker’s (source) voice to sound like another speaker (target), presents a threat to automatic speaker verification. In this paper, we first present new results of evaluating the vulnerability of current state-of-the-art speaker verification s ..."
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Cited by 11 (5 self)
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Abstract—Voice conversion technique, which modifies one speaker’s (source) voice to sound like another speaker (target), presents a threat to automatic speaker verification. In this paper, we first present new results of evaluating the vulnerability of current state-of-the-art speaker verification systems: Gaussian mixture model with joint factor analysis (GMM-JFA) and probabilistic linear discriminant analysis (PLDA) systems, against spoofing attacks. The spoofing attacks are simulated by two voice conversion techniques: Gaussian mixture model based conversion and unit selection based conversion. To reduce false acceptance rate caused by spoofing attack, we propose a general anti-spoofing attack framework for the speaker verification systems, where a converted speech detector is adopted as a post-processing module for the speaker verification system’s acceptance decision. The detector decides whether the accepted claim is human speech or converted speech. A subset of the core task in the NIST SRE 2006 corpus is used to evaluate the vulnerability of speaker verification system and the performance of converted speech detector. The results indicate that both conversion techniques can increase the false acceptance rate of GMM-JFA and PLDA system, while the converted speech detector can reduce the false acceptance rate
Exploiting intra-conversation variability for speaker diarization
- in INTERSPEECH
, 2011
"... In this paper, we propose a new approach to speaker diariza-tion based on the Total Variability approach to speaker verifica-tion. Drawing on previous work done in applying factor anal-ysis priors to the diarization problem, we arrive at a simplified approach that exploits intra-conversation variabi ..."
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Cited by 11 (2 self)
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In this paper, we propose a new approach to speaker diariza-tion based on the Total Variability approach to speaker verifica-tion. Drawing on previous work done in applying factor anal-ysis priors to the diarization problem, we arrive at a simplified approach that exploits intra-conversation variability in the To-tal Variability space through the use of Principal Component Analysis (PCA). Using our proposed methods, we demonstrate the ability to achieve state-of-the-art performance (0.9 % DER) in the diarization of summed-channel telephone data from the NIST 2008 SRE.
Patrol Team Language Identification System for DARPA RATS P1 Evaluation
"... This paper describes the language identification (LID) system developed by the Patrol team for the first phase of the DARPA RATS (Robust Automatic Transcription of Speech) program, which seeks to advance state of the art detection capabilities on audio from highly degraded communication channels. We ..."
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Cited by 10 (3 self)
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This paper describes the language identification (LID) system developed by the Patrol team for the first phase of the DARPA RATS (Robust Automatic Transcription of Speech) program, which seeks to advance state of the art detection capabilities on audio from highly degraded communication channels. We show that techniques originally developed for LID on telephone speech (e.g., for the NIST language recognition evaluations) remain effective on the noisy RATS data, provided that careful consideration is applied when designing the training and development sets. In addition, we show significant improvements from the use of Wiener filtering, neural network based and language dependent i-vector modeling, and fusion. Index Terms: language identification, noisy speech. 1.
AN INVESTIGATION ON BACK-END FOR SPEAKER RECOGNITION IN MULTI-SESSION ENROLLMENT
"... This study explores various back-end classifiers for robust speaker recognition in multi-session enrollment, with emphasis on optimal utilization and organization of speaker information present in the development data. Our objective is to construct a highly discriminative back-end framework by fusin ..."
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Cited by 10 (6 self)
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This study explores various back-end classifiers for robust speaker recognition in multi-session enrollment, with emphasis on optimal utilization and organization of speaker information present in the development data. Our objective is to construct a highly discriminative back-end framework by fusing several back-ends on an i-vector system framework. It is demonstrated that, by using different information/data configuration and modeling schemes, performance of the fused system can be significantly improved compared to an individual system using a single front-end and back-end. Averaged across both genders, we obtain a relative improvement in EER and minDCF by 56.5 % and 49.4%, respectively. Consistent performance gains obtained using the proposed strategy validates its effectiveness. This system is part of the CRSS ’ NIST SRE 2012 submission system.
Learning hidden unit contributions for unsupervised speaker adaptation of neural network acoustic models
- in Spoken Language Technology Workshop (SLT), 2014 IEEE. IEEE
"... This paper proposes a simple yet effective model-based neural network speaker adaptation technique that learns speaker-specific hidden unit contributions given adaptation data, without requiring any form of speaker-adaptive training, or labelled adaptation data. An additional amplitude parameter is ..."
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Cited by 10 (4 self)
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This paper proposes a simple yet effective model-based neural network speaker adaptation technique that learns speaker-specific hidden unit contributions given adaptation data, without requiring any form of speaker-adaptive training, or labelled adaptation data. An additional amplitude parameter is defined for each hidden unit; the amplitude parameters are tied for each speaker, and are learned using unsupervised adaptation. We conducted experiments on the TED talks data, as used in the International Workshop on Spoken Language Translation (IWSLT) evaluations. Our results indicate that the approach can reduce word error rates on standard IWSLT test sets by about 8–15 % relative compared to unadapted systems, with a further reduction of 4–6 % relative when combined with feature-space maximum likelihood linear re-gression (fMLLR). The approach can be employed in most existing feed-forward neural network architectures, and we report results using various hidden unit activation functions: sigmoid, maxout, and rectifying linear units (ReLU).
