1,245 research outputs found
Context-Dependent Acoustic Modeling without Explicit Phone Clustering
Phoneme-based acoustic modeling of large vocabulary automatic speech
recognition takes advantage of phoneme context. The large number of
context-dependent (CD) phonemes and their highly varying statistics require
tying or smoothing to enable robust training. Usually, Classification and
Regression Trees are used for phonetic clustering, which is standard in Hidden
Markov Model (HMM)-based systems. However, this solution introduces a secondary
training objective and does not allow for end-to-end training. In this work, we
address a direct phonetic context modeling for the hybrid Deep Neural Network
(DNN)/HMM, that does not build on any phone clustering algorithm for the
determination of the HMM state inventory. By performing different
decompositions of the joint probability of the center phoneme state and its
left and right contexts, we obtain a factorized network consisting of different
components, trained jointly. Moreover, the representation of the phonetic
context for the network relies on phoneme embeddings. The recognition accuracy
of our proposed models on the Switchboard task is comparable and outperforms
slightly the hybrid model using the standard state-tying decision trees.Comment: Submitted to Interspeech 202
Porting concepts from DNNs back to GMMs
Deep neural networks (DNNs) have been shown to outperform Gaussian Mixture Models (GMM) on a variety of speech recognition benchmarks. In this paper we analyze the differences between the DNN and GMM modeling techniques and port the best ideas from the DNN-based modeling to a GMM-based system. By going both deep (multiple layers) and wide (multiple parallel sub-models) and by sharing model parameters, we are able to close the gap between the two modeling techniques on the TIMIT database. Since the 'deep' GMMs retain the maximum-likelihood trained Gaussians as first layer, advanced techniques such as speaker adaptation and model-based noise robustness can be readily incorporated. Regardless of their similarities, the DNNs and the deep GMMs still show a sufficient amount of complementarity to allow effective system combination
On the Compression of Recurrent Neural Networks with an Application to LVCSR acoustic modeling for Embedded Speech Recognition
We study the problem of compressing recurrent neural networks (RNNs). In
particular, we focus on the compression of RNN acoustic models, which are
motivated by the goal of building compact and accurate speech recognition
systems which can be run efficiently on mobile devices. In this work, we
present a technique for general recurrent model compression that jointly
compresses both recurrent and non-recurrent inter-layer weight matrices. We
find that the proposed technique allows us to reduce the size of our Long
Short-Term Memory (LSTM) acoustic model to a third of its original size with
negligible loss in accuracy.Comment: Accepted in ICASSP 201
Improving large vocabulary continuous speech recognition by combining GMM-based and reservoir-based acoustic modeling
In earlier work we have shown that good phoneme recognition is possible with a so-called reservoir, a special type of recurrent neural network. In this paper, different architectures based on Reservoir Computing (RC) for large vocabulary continuous speech recognition are investigated. Besides experiments with HMM hybrids, it is shown that a RC-HMM tandem can achieve the same recognition accuracy as a classical HMM, which is a promising result for such a fairly new paradigm. It is also demonstrated that a state-level combination of the scores of the tandem and the baseline HMM leads to a significant improvement over the baseline. A word error rate reduction of the order of 20\% relative is possible
Advances in All-Neural Speech Recognition
This paper advances the design of CTC-based all-neural (or end-to-end) speech
recognizers. We propose a novel symbol inventory, and a novel iterated-CTC
method in which a second system is used to transform a noisy initial output
into a cleaner version. We present a number of stabilization and initialization
methods we have found useful in training these networks. We evaluate our system
on the commonly used NIST 2000 conversational telephony test set, and
significantly exceed the previously published performance of similar systems,
both with and without the use of an external language model and decoding
technology
Embedding-Based Speaker Adaptive Training of Deep Neural Networks
An embedding-based speaker adaptive training (SAT) approach is proposed and
investigated in this paper for deep neural network acoustic modeling. In this
approach, speaker embedding vectors, which are a constant given a particular
speaker, are mapped through a control network to layer-dependent element-wise
affine transformations to canonicalize the internal feature representations at
the output of hidden layers of a main network. The control network for
generating the speaker-dependent mappings is jointly estimated with the main
network for the overall speaker adaptive acoustic modeling. Experiments on
large vocabulary continuous speech recognition (LVCSR) tasks show that the
proposed SAT scheme can yield superior performance over the widely-used
speaker-aware training using i-vectors with speaker-adapted input features
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