Most current speech recognition systems use hidden Markov models (HMMs) to deal with the temporal variability of speech and Gaussian mixture models to determine how well each state of each HMM fits a frame or a short window of frames of coefficients that represents the acoustic input. An alternative way to evaluate the fit is to use a feed- forward neural network that takes several frames of coefficients as input and produces posterior probabilities over HMM states as output.

Deep neural networks with many hidden layers, that are trained using new methods have been shown to outperform Gaussian mixture models on a variety of speech recognition benchmarks, sometimes by a large margin. This paper provides an overview of this progress and represents the shared views of four research groups who have had recent successes in using deep neural networks for acoustic modeling in speech recognition.

**INTRODUCTION**

New machine learning algorithms can lead to significant advances in automatic speech recognition. The biggest single advance occured nearly four decades ago with the introduction of the Expectation-Maximization (EM) algorithm for training Hidden Markov Models (HMMs). With the EM algorithm, it became possible to develop speech recognition systems for real world tasks using the richness of Gaussian mixture models (GMM) to represent the relationship between HMM states and the acoustic input. In these systems the acoustic input is typically represented by concatenating Mel Frequency Cepstral Coefficients (MFCCs) or Perceptual Linear Predictive coefficients (PLPs) computed from the raw waveform, and their first- and second-order temporal differences. This non-adaptive but highly- engineered pre-processing of the waveform is designed to discard the large amount of information in waveforms that is considered to be irrelevant for discrimination and to express the remaining information in a form that facilitates discrimination with GMM-HMMs.

GMMs have a number of advantages that make them suitable for modeling the probability distributions over vectors of input features that are associated with each state of an HMM. With enough components, they can model probability distributions to any required level of accuracy and they are fairly easy to fit to data using the EM algorithm. A huge amount of research has gone into ways of constraining GMMs to increase their evaluation speed and to optimize the trade-off between their flexibility and the amount of training data available to avoid serious overfitting.

The recognition accuracy of a GMM-HMM system can be further improved if it is discriminatively fine-tuned after it has been generatively trained to maximize its probability of generating the observed data, especially if the discriminative objective function used for training is closely related to the error rate on phones, words or sentences[7]. The accuracy can also be improved by augmenting (or concatenating) the input features (e.g., MFCCs) with “tandem” or bottleneck features generated using neural networks. GMMs are so successful that it is difficult for any new method to outperform them for acoustic modeling.

Despite all their advantages, GMMs have a serious shortcoming – they are statistically inefficient for modeling data that lie on or near a non-linear manifold in the data space. For example, modeling the set of points that lie very close to the surface of a sphere only requires a few parameters using an appropriate model class, but it requires a very large number of diagonal Gaussians or a fairly large number of full-covariance Gaussians. Speech is produced by modulating a relatively small number of parameters of a dynamical system [10], [11] and this implies that its true underlying structure is much lower-dimensional than is immediately apparent in a window that contains hundreds of coefficients. We believe, therefore, that other types of model may work better than GMMs for acoustic modeling if they can more effectively exploit information embedded in a large window of frames.

Artificial neural networks trained by backpropagating error derivatives have the potential to learn much better models of data that lie on or near a non-linear manifold. In fact two decades ago, researchers achieved some success using artificial neural networks with a single layer of non-linear hidden units to predict HMM states from windows of acoustic coefficients. At that time, however, neither the hardware nor the learning algorithms were adequate for training neural networks with many hidden layers on large amounts of data and the performance benefits of using neural networks with a single hidden layer were not sufficiently large to seriously challenge GMMs. As a result, the main practical contribution of neural networks at that time was to provide extra features in tandem or bottleneck systems.

Over the last few years, advances in both machine learning algorithms and computer hardware have led to more efficient methods for training deep neural networks (DNNs) that contain many layers of non-linear hidden units and a very large output layer. The large output layer is required to accommodate the large number of HMM states that arise when each phone is modelled by a number of different “triphone” HMMs that take into account the phones on either side. Even when many of the states of these triphone HMMs are tied together, there can be thousands of tied states. Using the new learning methods, several different research groups have shown that DNNs can outperform GMMs at acoustic modeling for speech recognition on a variety of datasets including large datasets with large vocabularies.

This review paper aims to represent the shared views of research groups at the University of Toronto, Microsoft Research (MSR), Google and IBM Research, who have all had recent successes in using DNNs for acoustic modeling. The paper starts by describing the two-stage training procedure that is used for fitting the DNNs. In the first stage, layers of feature detectors are initialized, one layer at a time, by fitting a stack of generative models, each of which has one layer of latent variables. These generative models are trained without using any information about the HMM states that the acoustic model will need to discriminate. In the second stage, each generative model in the stack is used to initialize one layer of hidden units in a DNN and the whole network is then discriminatively fine-tuned to predict the target HMM states. These targets are obtained by using a baseline GMM-HMM system to produce a forced alignment.

In this paper we review exploratory experiments on the TIMIT database that were used to demonstrate the power of this two-stage training procedure for acoustic modeling. The DNNs that worked well on TIMIT were then applied to five different large vocabulary, continuous speech recognition tasks by three different research groups whose results we also summarize. The DNNs worked well on all of these tasks when compared with highly-tuned GMM-HMM systems and on some of the tasks they outperformed the state-of-the-art by a large margin. We also describe some other uses of DNNs for acoustic modeling and some variations on the training procedure.

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[Geoffrey Hinton, Li Deng, Dong Yu, Abdel-rahman Mohamed, Navdeep Jaitly, Andrew Senior, Vincent Vanhoucke, Patrick Nguyen, Tara Sainath, George Dahl, and Brian Kingsbury]