132 research outputs found
Robust language recognition via adaptive language factor extraction
This paper presents a technique to adapt an acoustically based
language classifier to the background conditions and speaker
accents. This adaptation improves language classification on
a broad spectrum of TV broadcasts. The core of the system
consists of an iVector-based setup in which language and channel
variabilities are modeled separately. The subsequent language
classifier (the backend) operates on the language factors,
i.e. those features in the extracted iVectors that explain the observed
language variability. The proposed technique adapts the
language variability model to the background conditions and
to the speaker accents present in the audio. The effect of the
adaptation is evaluated on a 28 hours corpus composed of documentaries and monolingual as well as multilingual broadcast
news shows. Consistent improvements in the automatic identification
of Flemish (Belgian Dutch), English and French are demonstrated for all broadcast types
Automatic speech recognition with deep neural networks for impaired speech
The final publication is available at https://link.springer.com/chapter/10.1007%2F978-3-319-49169-1_10Automatic Speech Recognition has reached almost human performance in some controlled scenarios. However, recognition of impaired speech is a difficult task for two main reasons: data is (i) scarce and (ii) heterogeneous. In this work we train different architectures on a database of dysarthric speech. A comparison between architectures shows that, even with a small database, hybrid DNN-HMM models outperform classical GMM-HMM according to word error rate measures. A DNN is able to improve the recognition word error rate a 13% for subjects with dysarthria with respect to the best classical architecture. This improvement is higher than the one given by other deep neural networks such as CNNs, TDNNs and LSTMs. All the experiments have been done with the Kaldi toolkit for speech recognition for which we have adapted several recipes to deal with dysarthric speech and work on the TORGO database. These recipes are publicly available.Peer ReviewedPostprint (author's final draft
Speaker recognition by means of Deep Belief Networks
Most state-of-the-art speaker recognition systems are based on Gaussian Mixture Models (GMMs), where a speech segment is represented by a compact representation, referred to as "identity vector" (ivector for short), extracted by means of Factor Analysis. The main advantage of this representation is that the problem of intersession variability is deferred to a second stage, dealing with low-dimensional vectors rather than with the high-dimensional space of the GMM means. In this paper, we propose to use as a pseudo-ivector extractor a Deep Belief Network (DBN) architecture, trained with the utterances of several hundred speakers. In this approach, the DBN performs a non-linear transformation of the input features, which produces the probability that an output unit is on, given the input features. We model the distribution of the output units, given an utterance, by a reduced set of parameters that embed the speaker characteristics. Tested on the dataset exploited for training the systems that have been used for the NIST 2012 Speaker Recognition Evaluation, this approach shows promising result
Speaker recognition by means of Deep Belief Networks
Most state–of–the–art speaker recognition systems are based on Gaussian Mixture Models (GMMs), where a speech segment is represented by a compact representation, referred to as “identity vector” (ivector for short), extracted by means of Factor Analysis. The main advantage of this
representation is that the problem of intersession variability is deferred to a second stage, dealing with low-dimensional vectors rather than with the high-dimensional space of the GMM means.
In this paper, we propose to use as a pseudo-ivector extractor a Deep Belief Network (DBN) architecture, trained with the utterances of several hundred speakers. In this approach, the DBN performs a non-linear
transformation of the input features, which produces the probability that an output unit is on, given the input features. We model the distribution of the output units, given an utterance, by a reduced set of parameters that
embed the speaker characteristics.
Tested on the dataset exploited for training the systems that have been used for the NIST 2012 Speaker Recognition Evaluation, this approach shows promising results
English Broadcast News Speech Recognition by Humans and Machines
With recent advances in deep learning, considerable attention has been given
to achieving automatic speech recognition performance close to human
performance on tasks like conversational telephone speech (CTS) recognition. In
this paper we evaluate the usefulness of these proposed techniques on broadcast
news (BN), a similar challenging task. We also perform a set of recognition
measurements to understand how close the achieved automatic speech recognition
results are to human performance on this task. On two publicly available BN
test sets, DEV04F and RT04, our speech recognition system using LSTM and
residual network based acoustic models with a combination of n-gram and neural
network language models performs at 6.5% and 5.9% word error rate. By achieving
new performance milestones on these test sets, our experiments show that
techniques developed on other related tasks, like CTS, can be transferred to
achieve similar performance. In contrast, the best measured human recognition
performance on these test sets is much lower, at 3.6% and 2.8% respectively,
indicating that there is still room for new techniques and improvements in this
space, to reach human performance levels.Comment: \copyright 2019 IEEE. Personal use of this material is permitted.
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Improving generalisation to new speakers in spoken dialogue state tracking
Users with disabilities can greatly benefit from personalised voice-enabled environmental-control interfaces, but for users with speech impairments (e.g. dysarthria) poor ASR performance poses a challenge to successful dialogue. Statistical dialogue management has shown resilience against high ASR error rates, hence making it useful to improve the performance of these interfaces. However, little research was devoted to dialogue management personalisation to specific users so far. Recently, data driven discriminative models have been shown to yield the best performance in dialogue state tracking (the inference of the user goal from the dialogue history). However, due to the unique characteristics of each speaker, training a system for a new user when user specific data is not available can be challenging due to the mismatch between training and working conditions. This work investigates two methods to improve the performance with new speakers of a LSTM-based personalised state tracker: The use of speaker specific acoustic and ASRrelated features; and dropout regularisation. It is shown that in an environmental control system for dysarthric speakers, the combination of both techniques yields improvements of 3.5% absolute in state tracking accuracy. Further analysis explores the effect of using different amounts of speaker specific data to train the tracking system
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Privacy-Preserving iVector-Based Speaker Verification
This paper introduces an efficient algorithm to develop a privacy-preserving voice verification based on iVector and linear discriminant analysis techniques. This research considers a scenario in which users enrol their voice biometric to access different services (i.e., banking). Once enrolment is completed, users can verify themselves using their voice print instead of alphanumeric passwords. Since a voice print is unique for everyone, storing it with a third-party server raises several privacy concerns. To address this challenge, this paper proposes a novel technique based on randomization to carry out voice authentication, which allows the user to enrol and verify their voice in the randomized domain. To achieve this, the iVector-based voice verification technique has been redesigned to work on the randomized domain. The proposed algorithm is validated using a well-known speech dataset. The proposed algorithm neither compromises the authentication accuracy nor adds additional complexity due to the randomization operations
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