15,739 research outputs found
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
Robust ASR using Support Vector Machines
The improved theoretical properties of Support Vector Machines with respect to other machine learning alternatives due to their max-margin training paradigm have led us to suggest them as a good technique for robust speech recognition. However, important shortcomings have had to be circumvented, the most important being the normalisation of the time duration of different realisations of the acoustic speech units.
In this paper, we have compared two approaches in noisy environments: first, a hybrid HMM–SVM solution where a fixed number of frames is selected by means of an HMM segmentation and second, a normalisation kernel called Dynamic Time Alignment Kernel (DTAK) first introduced in Shimodaira et al. [Shimodaira, H., Noma, K., Nakai, M., Sagayama, S., 2001. Support vector machine with dynamic time-alignment kernel for speech recognition. In: Proc. Eurospeech, Aalborg, Denmark, pp. 1841–1844] and based on DTW (Dynamic Time Warping). Special attention has been paid to the adaptation of both alternatives to noisy environments, comparing two types of parameterisations and performing suitable feature normalisation operations. The results show that the DTA Kernel provides important advantages over the baseline HMM system in medium to bad noise conditions, also outperforming the results of the hybrid system.Publicad
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
A Subband-Based SVM Front-End for Robust ASR
This work proposes a novel support vector machine (SVM) based robust
automatic speech recognition (ASR) front-end that operates on an ensemble of
the subband components of high-dimensional acoustic waveforms. The key issues
of selecting the appropriate SVM kernels for classification in frequency
subbands and the combination of individual subband classifiers using ensemble
methods are addressed. The proposed front-end is compared with state-of-the-art
ASR front-ends in terms of robustness to additive noise and linear filtering.
Experiments performed on the TIMIT phoneme classification task demonstrate the
benefits of the proposed subband based SVM front-end: it outperforms the
standard cepstral front-end in the presence of noise and linear filtering for
signal-to-noise ratio (SNR) below 12-dB. A combination of the proposed
front-end with a conventional front-end such as MFCC yields further
improvements over the individual front ends across the full range of noise
levels
Articulatory and bottleneck features for speaker-independent ASR of dysarthric speech
The rapid population aging has stimulated the development of assistive
devices that provide personalized medical support to the needies suffering from
various etiologies. One prominent clinical application is a computer-assisted
speech training system which enables personalized speech therapy to patients
impaired by communicative disorders in the patient's home environment. Such a
system relies on the robust automatic speech recognition (ASR) technology to be
able to provide accurate articulation feedback. With the long-term aim of
developing off-the-shelf ASR systems that can be incorporated in clinical
context without prior speaker information, we compare the ASR performance of
speaker-independent bottleneck and articulatory features on dysarthric speech
used in conjunction with dedicated neural network-based acoustic models that
have been shown to be robust against spectrotemporal deviations. We report ASR
performance of these systems on two dysarthric speech datasets of different
characteristics to quantify the achieved performance gains. Despite the
remaining performance gap between the dysarthric and normal speech, significant
improvements have been reported on both datasets using speaker-independent ASR
architectures.Comment: to appear in Computer Speech & Language -
https://doi.org/10.1016/j.csl.2019.05.002 - arXiv admin note: substantial
text overlap with arXiv:1807.1094
Improved training of end-to-end attention models for speech recognition
Sequence-to-sequence attention-based models on subword units allow simple
open-vocabulary end-to-end speech recognition. In this work, we show that such
models can achieve competitive results on the Switchboard 300h and LibriSpeech
1000h tasks. In particular, we report the state-of-the-art word error rates
(WER) of 3.54% on the dev-clean and 3.82% on the test-clean evaluation subsets
of LibriSpeech. We introduce a new pretraining scheme by starting with a high
time reduction factor and lowering it during training, which is crucial both
for convergence and final performance. In some experiments, we also use an
auxiliary CTC loss function to help the convergence. In addition, we train long
short-term memory (LSTM) language models on subword units. By shallow fusion,
we report up to 27% relative improvements in WER over the attention baseline
without a language model.Comment: submitted to Interspeech 201
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