21 research outputs found
Transfer Learning for Speech and Language Processing
Transfer learning is a vital technique that generalizes models trained for
one setting or task to other settings or tasks. For example in speech
recognition, an acoustic model trained for one language can be used to
recognize speech in another language, with little or no re-training data.
Transfer learning is closely related to multi-task learning (cross-lingual vs.
multilingual), and is traditionally studied in the name of `model adaptation'.
Recent advance in deep learning shows that transfer learning becomes much
easier and more effective with high-level abstract features learned by deep
models, and the `transfer' can be conducted not only between data distributions
and data types, but also between model structures (e.g., shallow nets and deep
nets) or even model types (e.g., Bayesian models and neural models). This
review paper summarizes some recent prominent research towards this direction,
particularly for speech and language processing. We also report some results
from our group and highlight the potential of this very interesting research
field.Comment: 13 pages, APSIPA 201
Learning representations for speech recognition using artificial neural networks
Learning representations is a central challenge in machine learning. For speech
recognition, we are interested in learning robust representations that are stable
across different acoustic environments, recording equipment and irrelevant inter–
and intra– speaker variabilities. This thesis is concerned with representation
learning for acoustic model adaptation to speakers and environments, construction
of acoustic models in low-resource settings, and learning representations from
multiple acoustic channels. The investigations are primarily focused on the hybrid
approach to acoustic modelling based on hidden Markov models and artificial
neural networks (ANN).
The first contribution concerns acoustic model adaptation. This comprises
two new adaptation transforms operating in ANN parameters space. Both operate
at the level of activation functions and treat a trained ANN acoustic model as
a canonical set of fixed-basis functions, from which one can later derive variants
tailored to the specific distribution present in adaptation data. The first technique,
termed Learning Hidden Unit Contributions (LHUC), depends on learning
distribution-dependent linear combination coefficients for hidden units. This
technique is then extended to altering groups of hidden units with parametric and
differentiable pooling operators. We found the proposed adaptation techniques
pose many desirable properties: they are relatively low-dimensional, do not overfit
and can work in both a supervised and an unsupervised manner. For LHUC we
also present extensions to speaker adaptive training and environment factorisation.
On average, depending on the characteristics of the test set, 5-25% relative
word error rate (WERR) reductions are obtained in an unsupervised two-pass
adaptation setting.
The second contribution concerns building acoustic models in low-resource
data scenarios. In particular, we are concerned with insufficient amounts of
transcribed acoustic material for estimating acoustic models in the target language
– thus assuming resources like lexicons or texts to estimate language models
are available. First we proposed an ANN with a structured output layer
which models both context–dependent and context–independent speech units,
with the context-independent predictions used at runtime to aid the prediction
of context-dependent states. We also propose to perform multi-task adaptation
with a structured output layer. We obtain consistent WERR reductions up to
6.4% in low-resource speaker-independent acoustic modelling. Adapting those
models in a multi-task manner with LHUC decreases WERRs by an additional
13.6%, compared to 12.7% for non multi-task LHUC. We then demonstrate that
one can build better acoustic models with unsupervised multi– and cross– lingual
initialisation and find that pre-training is a largely language-independent. Up to
14.4% WERR reductions are observed, depending on the amount of the available
transcribed acoustic data in the target language.
The third contribution concerns building acoustic models from multi-channel
acoustic data. For this purpose we investigate various ways of integrating and
learning multi-channel representations. In particular, we investigate channel concatenation
and the applicability of convolutional layers for this purpose. We
propose a multi-channel convolutional layer with cross-channel pooling, which
can be seen as a data-driven non-parametric auditory attention mechanism. We
find that for unconstrained microphone arrays, our approach is able to match the
performance of the comparable models trained on beamform-enhanced signals
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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
Automatic Speech Recognition without Transcribed Speech or Pronunciation Lexicons
Rapid deployment of automatic speech recognition (ASR) in new languages, with very limited data, is of great interest and importance for intelligence gathering, as well as for humanitarian assistance and disaster relief (HADR). Deploying ASR systems in these languages often relies on cross-lingual acoustic modeling followed by supervised adaptation and almost always assumes that either a pronunciation lexicon using the International Phonetic Alphabet (IPA), and/or some amount of transcribed speech exist in the new language of interest. For many languages, neither requirement is generally true -- only a limited amount of text and untranscribed audio is available. This work focuses specifically on scalable techniques for building ASR systems in most languages without any existing transcribed speech or pronunciation lexicons.
We first demonstrate how cross-lingual acoustic model transfer, when phonemic pronunciation lexicons do exist in a new language, can significantly reduce the need for target-language transcribed speech. We then explore three methods for handling languages without a pronunciation lexicon. First we examine the effectiveness of graphemic acoustic model transfer, which allows for pronunciation lexicons to be trivially constructed. We then present two methods for rapid construction of phonemic pronunciation lexicons based on submodular selection of a small set of words for manual annotation, or words from other languages for which we have IPA pronunciations. We also explore techniques for training sequence-to-sequence models with very small amounts of data by transferring models trained on other languages, and leveraging large unpaired text corpora in training. Finally, as an alternative to acoustic model transfer, we present a novel hybrid generative/discriminative semi-supervised training framework that merges recent progress in Energy Based Models (EBMs) as well as lattice-free maximum mutual information (LF-MMI) training, capable of making use of purely untranscribed audio.
Together, these techniques enabled ASR capabilities that supported triage of spoken communications in real-world HADR work-flows in many languages using fewer than 30 minutes of transcribed speech. These techniques were successfully applied in multiple NIST evaluations and were among the top-performing systems in each evaluation
Code-Switched Urdu ASR for Noisy Telephonic Environment using Data Centric Approach with Hybrid HMM and CNN-TDNN
Call Centers have huge amount of audio data which can be used for achieving
valuable business insights and transcription of phone calls is manually tedious
task. An effective Automated Speech Recognition system can accurately
transcribe these calls for easy search through call history for specific
context and content allowing automatic call monitoring, improving QoS through
keyword search and sentiment analysis. ASR for Call Center requires more
robustness as telephonic environment are generally noisy. Moreover, there are
many low-resourced languages that are on verge of extinction which can be
preserved with help of Automatic Speech Recognition Technology. Urdu is the
most widely spoken language in the world, with 231,295,440 worldwide
still remains a resource constrained language in ASR. Regional call-center
conversations operate in local language, with a mix of English numbers and
technical terms generally causing a "code-switching" problem. Hence, this paper
describes an implementation framework of a resource efficient Automatic Speech
Recognition/ Speech to Text System in a noisy call-center environment using
Chain Hybrid HMM and CNN-TDNN for Code-Switched Urdu Language. Using Hybrid
HMM-DNN approach allowed us to utilize the advantages of Neural Network with
less labelled data. Adding CNN with TDNN has shown to work better in noisy
environment due to CNN's additional frequency dimension which captures extra
information from noisy speech, thus improving accuracy. We collected data from
various open sources and labelled some of the unlabelled data after analysing
its general context and content from Urdu language as well as from commonly
used words from other languages, primarily English and were able to achieve WER
of 5.2% with noisy as well as clean environment in isolated words or numbers as
well as in continuous spontaneous speech.Comment: 32 pages, 19 figures, 2 tables, preprin
Deep representation learning for speech recognition
Representation learning is a fundamental ingredient of deep learning. However, learning a good representation is a challenging task. For speech recognition, such a representation should contain the information needed to perform well in this task. A robust representation should also be reusable, hence it should capture the structure of the data. Interpretability is another desired characteristic. In this thesis we strive to learn an optimal deep representation for speech recognition using feed-forward Neural Networks (NNs) with different connectivity patterns.
First and foremost, we aim to improve the robustness of the acoustic models. We use attribute-aware and adaptive training strategies to model the underlying factors of variation related to the speakers and the acoustic conditions. We focus on low-latency and real-time decoding scenarios. We explore different utterance summaries (referred to as utterance embeddings), capturing various sources of speech variability, and we seek to optimise speaker adaptive training (SAT) with control networks acting on the embeddings. We also propose a multi-scale CNN layer, to learn factorised representations. The proposed multi-scale approach also tackles the computational and memory efficiency.
We also present a number of different approaches as an attempt to better understand learned representations. First, with a controlled design, we aim to assess the role of individual components of deep CNN acoustic models. Next, with saliency maps, we evaluate the importance of each input feature with respect to the classification criterion. Then, we propose to evaluate layer-wise and model-wise learned representations in different diagnostic verification tasks (speaker and acoustic condition verification). We propose a deep CNN model as the embedding extractor, merging the information learned at different layers in the network. Similarly, we perform the analyses for the embeddings used in SAT-DNNs to gain more insight. For the multi-scale models, we also show how to compare learned representations (and assess their robustness) with a metric invariant to affine transformations