11 research outputs found
Optimizing expected word error rate via sampling for speech recognition
State-level minimum Bayes risk (sMBR) training has become the de facto
standard for sequence-level training of speech recognition acoustic models. It
has an elegant formulation using the expectation semiring, and gives large
improvements in word error rate (WER) over models trained solely using
cross-entropy (CE) or connectionist temporal classification (CTC). sMBR
training optimizes the expected number of frames at which the reference and
hypothesized acoustic states differ. It may be preferable to optimize the
expected WER, but WER does not interact well with the expectation semiring, and
previous approaches based on computing expected WER exactly involve expanding
the lattices used during training. In this paper we show how to perform
optimization of the expected WER by sampling paths from the lattices used
during conventional sMBR training. The gradient of the expected WER is itself
an expectation, and so may be approximated using Monte Carlo sampling. We show
experimentally that optimizing WER during acoustic model training gives 5%
relative improvement in WER over a well-tuned sMBR baseline on a 2-channel
query recognition task (Google Home)
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Optimisation Methods For Training Deep Neural Networks in Speech Recognition
Automatic Speech Recognition (ASR) is an example of a sequence to sequence level classification task where, given an acoustic waveform, the goal is to produce the correct word level hypotheses. In machine learning, a classification problem such as ASR is solved in two stages: an inference stage that models the uncertainty associated with the choice of hypothesis given the acoustic waveform using a mathematical model, and a decision stage which employs the inference model in conjunction with decision theory to make optimal class assignments. With the advent of careful network initialisation and GPU computing, hybrid Hidden Markov Models (HMMs) augmented with Deep Neural Networks (DNNs) have shown to outperform traditional HMMs using Gaussian Mixture Models (GMMs) in solving the inference problem for ASR. In comparison to GMMs, DNNs possess a better capability to model the underlying non-linear data manifold due to their deep and complex structure. While the structure of such models gives rich modelling capability, it also creates complex dependencies between the parameters which can make learning difficult via first order stochastic gradient descent (SGD). The task of finding the best procedure to train DNNs continues to be an active area of research and has been made even more challenging by the availability of ever more training data. This thesis focuses on designing better optimisation approaches to train hybrid HMM-DNN models using sequence level discriminative criterion which is a natural loss function that preserves the sequential ordering of frames within a spoken utterance. The thesis presents an implementation of the second order Hessian Free (HF) optimisation method, and shows how the method can made efficient through appropriate modifications to the Conjugate Gradient algorithm. To achieve better convergence than SGD, this work explores the Natural Gradient method to train DNNs with discriminative sequence training. In the DNN literature, the method has been applied to train models for the Maximum Likelihood objective criterion. A novel contribution of this thesis is to extend this approach to the domain of Minimum Bayes Risk objective functions for discriminative sequence training. With sigmoid models trained on a 50hr and 200hr training set from the Multi-Genre Broadcast 1 (MGB1) transcription task, the NG method applied in a HF styled optimisation framework is shown to achieve better Word Error Rate (WER) reductions on the MGB1 development set than SGD from sequence training.
This thesis also addresses the particular issue of overfitting between the training criterion and WER, that primarily arises during sequence training of DNN models that use Rectified Linear Units (ReLUs) as activation functions. It is shown how by scaling with the Gauss Newton matrix, the HF method unlike other approaches can overcome this issue. Seeing that different optimisers work best with different models, it is attractive to have a consistent optimisation framework that is agnostic to the choice of activation function. To address the issue, this thesis develops the geometry of the underlying function space captured by different realisations of DNN model parameters, and presents the design considerations for an optimisation algorithm to be well defined on this space. Building on this analysis, a novel optimisation technique called NGHF is presented that uses both the direction of steepest descent on a probabilistic manifold and local curvature information to effectively probe the error surface. The basis of the method relies on an alternative derivation of Taylorâs theorem using the concepts of manifolds, tangent vectors and directional derivatives from the perspective of Information Geometry. Apart from being well defined on the function space, when framed within a HF style optimisation framework, the method of NGHF is shown to achieve the greatest WER reductions from sequence training on the MGB1 development set with both sigmoid and ReLU based models trained on the 200hr MGB1 training set. The evaluation of the above optimisation methods in training different DNN model architectures is also presented.IDB Cambridge International Scholarshi
Full Covariance Modelling for Speech Recognition
HMM-based systems for Automatic Speech Recognition typically model
the acoustic features using mixtures of multivariate Gaussians. In this
thesis, we consider the problem of learning a suitable covariance matrix
for each Gaussian. A variety of schemes have been proposed for
controlling the number of covariance parameters per Gaussian, and
studies have shown that in general, the greater the number of parameters
used in the models, the better the recognition performance. We
therefore investigate systems with full covariance Gaussians. However,
in this case, the obvious choice of parameters â given by the sample
covariance matrix â leads to matrices that are poorly-conditioned, and
do not generalise well to unseen test data. The problem is particularly
acute when the amount of training data is limited.
We propose two solutions to this problem: firstly, we impose the requirement
that each matrix should take the form of a Gaussian graphical
model, and introduce a method for learning the parameters and
the model structure simultaneously. Secondly, we explain how an
alternative estimator, the shrinkage estimator, is preferable to the
standard maximum likelihood estimator, and derive formulae for the
optimal shrinkage intensity within the context of a Gaussian mixture
model. We show how this relates to the use of a diagonal covariance
smoothing prior.
We compare the effectiveness of these techniques to standard methods
on a phone recognition task where the quantity of training data is
artificially constrained. We then investigate the performance of the
shrinkage estimator on a large-vocabulary conversational telephone
speech recognition task. Discriminative training techniques can be used to compensate for the
invalidity of the model correctness assumption underpinning maximum
likelihood estimation. On the large-vocabulary task, we use discriminative
training of the full covariance models and diagonal priors
to yield improved recognition performance
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Joint Training Methods for Tandem and Hybrid Speech Recognition Systems using Deep Neural Networks
Hidden Markov models (HMMs) have been the mainstream acoustic modelling approach for state-of-the-art automatic speech recognition (ASR) systems over the
past few decades. Recently, due to the rapid development of deep learning technologies, deep neural networks (DNNs) have become an essential part of nearly all kinds of ASR approaches. Among HMM-based ASR approaches, DNNs are most commonly used to extract features (tandem system configuration) or to directly produce HMM output probabilities (hybrid system configuration).
Although DNN tandem and hybrid systems have been shown to have superior
performance to traditional ASR systems without any DNN models, there are still
issues with such systems. First, some of the DNN settings, such as the choice of
the context-dependent (CD) output targets set and hidden activation functions, are
usually determined independently from the DNN training process. Second, different
ASR modules are separately optimised based on different criteria following a greedy
build strategy. For instance, for tandem systems, the features are often extracted by a
DNN trained to classify individual speech frames while acoustic models are built upon
such features according to a sequence level criterion. These issues mean that the best performance is not theoretically guaranteed.
This thesis focuses on alleviating both issues using joint training methods. In DNN
acoustic model joint training, the decision tree HMM state tying approach is extended
to cluster DNN-HMM states. Based on this method, an alternative CD-DNN training
procedure without relying on any additional system is proposed, which can produce
DNN acoustic models comparable in word error rate (WER) with those trained by the
conventional procedure. Meanwhile, the most common hidden activation functions,
the sigmoid and rectified linear unit (ReLU), are parameterised to enable automatic
learning of function forms. Experiments using conversational telephone speech (CTS)
Mandarin data result in an average of 3.4% and 2.2% relative character error rate (CER) reduction with sigmoid and ReLU parameterisations. Such parameterised functions can also be applied to speaker adaptation tasks.
At the ASR system level, DNN acoustic model and corresponding speaker dependent (SD) input feature transforms are jointly learned through minimum phone error
(MPE) training as an example of hybrid system joint training, which outperforms the
conventional hybrid system speaker adaptive training (SAT) method. MPE based speaker independent (SI) tandem system joint training is also studied. Experiments on
multi-genre broadcast (MGB) English data show that this method gives a reduction
in tandem system WER of 11.8% (relative), and the resulting tandem systems are
comparable to MPE hybrid systems in both WER and the number of parameters. In
addition, all approaches in this thesis have been implemented using the hidden Markov model toolkit (HTK) and the related source code has been or will be made publicly available with either recent or future HTK releases, to increase the reproducibility of the work presented in this thesis.Cambridge International Scholarship, Cambridge Overseas Trust
Research funding, EPSRC Natural Speech Technology Project
Research funding, DARPA BOLT Program
Research funding, iARPA Babel Progra
Pinched Lattice Minimum Bayes Risk Discriminative Training for Large Vocabulary Continuous Speech Recognition
Iterative estimation procedures that minimize empirical risk based on general loss functions such as the Levenshtein distance have been derived as extensions of the Extended Baum Welch algorithm. While reducing expected loss on training data is a desirable training criterion, these algorithms can be difficult to apply. They are unlike MMI estimation in that they require an explicit listing of the hypotheses to be considered and in complex problems such lists tend to be prohibitively large. To overcome this difficulty, modeling techniques originally developed to improve search efficiency in Minimum Bayes Risk decoding can be used to transform these estimation algorithms so that exact update, risk minimization procedures can be used for complex recognition problems. Experimental results in two large vocabulary speech recognition tasks show improvements over conventionally trained MMIE models