1,996 research outputs found
Speaker verification using sequence discriminant support vector machines
This paper presents a text-independent speaker verification system using support vector machines (SVMs) with score-space kernels. Score-space kernels generalize Fisher kernels and are based on underlying generative models such as Gaussian mixture models (GMMs). This approach provides direct discrimination between whole sequences, in contrast with the frame-level approaches at the heart of most current systems. The resultant SVMs have a very high dimensionality since it is related to the number of parameters in the underlying generative model. To address problems that arise in the resultant optimization we introduce a technique called spherical normalization that preconditions the Hessian matrix. We have performed speaker verification experiments using the PolyVar database. The SVM system presented here reduces the relative error rates by 34% compared to a GMM likelihood ratio system
Semi-supervised transductive speaker identification
We present an application of transductive semi-supervised learning to the problem of speaker identification. Formulating this problem as one of transduction is the most natural choice in some scenarios, such as when annotating archived speech data. Experiments with the CHAINS corpus show that, using the basic MFCC-encoding of recorded utterances, a well known simple semi-supervised algorithm, label spread, can solve this problem well. With only a small number of labelled utterances, the semi-supervised algorithm drastically outperforms a state of the art supervised support vector machine algorithm. Although we restrict ourselves to the transductive setting in this paper, the results encourage future work on semi-supervised learning for inductive speaker identification
Exploring the Encoding Layer and Loss Function in End-to-End Speaker and Language Recognition System
In this paper, we explore the encoding/pooling layer and loss function in the
end-to-end speaker and language recognition system. First, a unified and
interpretable end-to-end system for both speaker and language recognition is
developed. It accepts variable-length input and produces an utterance level
result. In the end-to-end system, the encoding layer plays a role in
aggregating the variable-length input sequence into an utterance level
representation. Besides the basic temporal average pooling, we introduce a
self-attentive pooling layer and a learnable dictionary encoding layer to get
the utterance level representation. In terms of loss function for open-set
speaker verification, to get more discriminative speaker embedding, center loss
and angular softmax loss is introduced in the end-to-end system. Experimental
results on Voxceleb and NIST LRE 07 datasets show that the performance of
end-to-end learning system could be significantly improved by the proposed
encoding layer and loss function.Comment: Accepted for Speaker Odyssey 201
- ā¦