4,248 research outputs found
Latent Class Model with Application to Speaker Diarization
In this paper, we apply a latent class model (LCM) to the task of speaker
diarization. LCM is similar to Patrick Kenny's variational Bayes (VB) method in
that it uses soft information and avoids premature hard decisions in its
iterations. In contrast to the VB method, which is based on a generative model,
LCM provides a framework allowing both generative and discriminative models.
The discriminative property is realized through the use of i-vector (Ivec),
probabilistic linear discriminative analysis (PLDA), and a support vector
machine (SVM) in this work. Systems denoted as LCM-Ivec-PLDA, LCM-Ivec-SVM, and
LCM-Ivec-Hybrid are introduced. In addition, three further improvements are
applied to enhance its performance. 1) Adding neighbor windows to extract more
speaker information for each short segment. 2) Using a hidden Markov model to
avoid frequent speaker change points. 3) Using an agglomerative hierarchical
cluster to do initialization and present hard and soft priors, in order to
overcome the problem of initial sensitivity. Experiments on the National
Institute of Standards and Technology Rich Transcription 2009 speaker
diarization database, under the condition of a single distant microphone, show
that the diarization error rate (DER) of the proposed methods has substantial
relative improvements compared with mainstream systems. Compared to the VB
method, the relative improvements of LCM-Ivec-PLDA, LCM-Ivec-SVM, and
LCM-Ivec-Hybrid systems are 23.5%, 27.1%, and 43.0%, respectively. Experiments
on our collected database, CALLHOME97, CALLHOME00 and SRE08 short2-summed trial
conditions also show that the proposed LCM-Ivec-Hybrid system has the best
overall performance
Comparing Human and Machine Errors in Conversational Speech Transcription
Recent work in automatic recognition of conversational telephone speech (CTS)
has achieved accuracy levels comparable to human transcribers, although there
is some debate how to precisely quantify human performance on this task, using
the NIST 2000 CTS evaluation set. This raises the question what systematic
differences, if any, may be found differentiating human from machine
transcription errors. In this paper we approach this question by comparing the
output of our most accurate CTS recognition system to that of a standard speech
transcription vendor pipeline. We find that the most frequent substitution,
deletion and insertion error types of both outputs show a high degree of
overlap. The only notable exception is that the automatic recognizer tends to
confuse filled pauses ("uh") and backchannel acknowledgments ("uhhuh"). Humans
tend not to make this error, presumably due to the distinctive and opposing
pragmatic functions attached to these words. Furthermore, we quantify the
correlation between human and machine errors at the speaker level, and
investigate the effect of speaker overlap between training and test data.
Finally, we report on an informal "Turing test" asking humans to discriminate
between automatic and human transcription error cases
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
Generative Modelling for Unsupervised Score Calibration
Score calibration enables automatic speaker recognizers to make
cost-effective accept / reject decisions. Traditional calibration requires
supervised data, which is an expensive resource. We propose a 2-component GMM
for unsupervised calibration and demonstrate good performance relative to a
supervised baseline on NIST SRE'10 and SRE'12. A Bayesian analysis demonstrates
that the uncertainty associated with the unsupervised calibration parameter
estimates is surprisingly small.Comment: Accepted for ICASSP 201
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