63 research outputs found
Advances in deep learning methods for speech recognition and understanding
Ce travail expose plusieurs études dans les domaines de
la reconnaissance de la parole et
compréhension du langage parlé.
La compréhension sémantique du langage parlé est un sous-domaine important
de l'intelligence artificielle.
Le traitement de la parole intéresse depuis longtemps les chercheurs,
puisque la parole est une des charactéristiques qui definit l'être humain.
Avec le développement du réseau neuronal artificiel,
le domaine a connu une évolution rapide
à la fois en terme de précision et de perception humaine.
Une autre étape importante a été franchie avec le développement
d'approches bout en bout.
De telles approches permettent une coadaptation de toutes
les parties du modèle, ce qui augmente ainsi les performances,
et ce qui simplifie la procédure d'entrainement.
Les modèles de bout en bout sont devenus réalisables avec la quantité croissante
de données disponibles, de ressources informatiques et,
surtout, avec de nombreux développements architecturaux innovateurs.
Néanmoins, les approches traditionnelles (qui ne sont pas bout en bout)
sont toujours pertinentes pour le traitement de la parole en raison
des données difficiles dans les environnements bruyants,
de la parole avec un accent et de la grande variété de dialectes.
Dans le premier travail, nous explorons la reconnaissance de la parole hybride
dans des environnements bruyants.
Nous proposons de traiter la reconnaissance de la parole,
qui fonctionne dans
un nouvel environnement composé de différents bruits inconnus,
comme une tâche d'adaptation de domaine.
Pour cela, nous utilisons la nouvelle technique à l'époque
de l'adaptation du domaine antagoniste.
En résumé, ces travaux antérieurs proposaient de former
des caractéristiques de manière à ce qu'elles soient distinctives
pour la tâche principale, mais non-distinctive pour la tâche secondaire.
Cette tâche secondaire est conçue pour être la tâche de reconnaissance de domaine.
Ainsi, les fonctionnalités entraînées sont invariantes vis-à-vis du domaine considéré.
Dans notre travail, nous adoptons cette technique et la modifions pour
la tâche de reconnaissance de la parole dans un environnement bruyant.
Dans le second travail, nous développons une méthode générale
pour la régularisation des réseaux génératif récurrents.
Il est connu que les réseaux récurrents ont souvent des difficultés à rester
sur le même chemin, lors de la production de sorties longues.
Bien qu'il soit possible d'utiliser des réseaux bidirectionnels pour
une meilleure traitement de séquences pour l'apprentissage des charactéristiques,
qui n'est pas applicable au cas génératif.
Nous avons développé un moyen d'améliorer la cohérence de
la production de longues séquences avec des réseaux récurrents.
Nous proposons un moyen de construire un modèle similaire à un réseau bidirectionnel.
L'idée centrale est d'utiliser une perte L2 entre
les réseaux récurrents génératifs vers l'avant et vers l'arrière.
Nous fournissons une évaluation expérimentale sur
une multitude de tâches et d'ensembles de données,
y compris la reconnaissance vocale,
le sous-titrage d'images et la modélisation du langage.
Dans le troisième article, nous étudions la possibilité de développer
un identificateur d'intention de bout en bout pour la compréhension du langage parlé.
La compréhension sémantique du langage parlé est une étape importante vers
le développement d'une intelligence artificielle de type humain.
Nous avons vu que les approches de bout en bout montrent
des performances élevées sur les tâches, y compris la traduction automatique et
la reconnaissance de la parole.
Nous nous inspirons des travaux antérieurs pour développer
un système de bout en bout pour la reconnaissance de l'intention.This work presents several studies in the areas of speech recognition and
understanding.
The semantic speech understanding is an important sub-domain of the
broader field of artificial intelligence.
Speech processing has had interest from the researchers for long time
because language is one of the defining characteristics of a human being.
With the development of neural networks, the domain has seen rapid progress
both in terms of accuracy and human perception.
Another important milestone was achieved with the development of
end-to-end approaches.
Such approaches allow co-adaptation of all the parts of the model
thus increasing the performance, as well as simplifying the training
procedure.
End-to-end models became feasible with the increasing amount of available
data, computational resources, and most importantly with many novel
architectural developments.
Nevertheless, traditional, non end-to-end, approaches are still relevant
for speech processing due to challenging data in noisy environments,
accented speech, and high variety of dialects.
In the first work, we explore the hybrid speech recognition in noisy
environments.
We propose to treat the recognition in the unseen noise condition
as the domain adaptation task.
For this, we use the novel at the time technique of the adversarial
domain adaptation.
In the nutshell, this prior work proposed to train features in such
a way that they are discriminative for the primary task,
but non-discriminative for the secondary task.
This secondary task is constructed to be the domain recognition task.
Thus, the features trained are invariant towards the domain at hand.
In our work, we adopt this technique and modify it for the task of
noisy speech recognition.
In the second work, we develop a general method for regularizing
the generative recurrent networks.
It is known that the recurrent networks frequently have difficulties
staying on same track when generating long outputs.
While it is possible to use bi-directional networks for better
sequence aggregation for feature learning, it is not applicable
for the generative case.
We developed a way improve the consistency of generating long sequences
with recurrent networks.
We propose a way to construct a model similar to bi-directional network.
The key insight is to use a soft L2 loss between the forward and
the backward generative recurrent networks.
We provide experimental evaluation on a multitude of tasks and datasets,
including speech recognition, image captioning, and language modeling.
In the third paper, we investigate the possibility of developing
an end-to-end intent recognizer for spoken language understanding.
The semantic spoken language understanding is an important
step towards developing a human-like artificial intelligence.
We have seen that the end-to-end approaches show high
performance on the tasks including machine translation and speech recognition.
We draw the inspiration from the prior works to develop
an end-to-end system for intent recognition
Spoken command recognition for robotics
In this thesis, I investigate spoken command recognition technology for robotics. While high
robustness is expected, the distant and noisy conditions in which the system has to operate
make the task very challenging. Unlike commercial systems which all rely on a "wake-up"
word to initiate the interaction, the pipeline proposed here directly detect and recognizes
commands from the continuous audio stream. In order to keep the task manageable despite
low-resource conditions, I propose to focus on a limited set of commands, thus trading off
flexibility of the system against robustness.
Domain and speaker adaptation strategies based on a multi-task regularization paradigm
are first explored. More precisely, two different methods are proposed which rely on a tied
loss function which penalizes the distance between the output of several networks. The first
method considers each speaker or domain as a task. A canonical task-independent network is
jointly trained with task-dependent models, allowing both types of networks to improve by
learning from one another. While an improvement of 3.2% on the frame error rate (FER) of
the task-independent network is obtained, this only partially carried over to the phone error
rate (PER), with 1.5% of improvement. Similarly, a second method explored the parallel
training of the canonical network with a privileged model having access to i-vectors. This
method proved less effective with only 1.2% of improvement on the FER.
In order to make the developed technology more accessible, I also investigated the use
of a sequence-to-sequence (S2S) architecture for command classification. The use of an
attention-based encoder-decoder model reduced the classification error by 40% relative to a
strong convolutional neural network (CNN)-hidden Markov model (HMM) baseline, showing
the relevance of S2S architectures in such context. In order to improve the flexibility of the
trained system, I also explored strategies for few-shot learning, which allow to extend the
set of commands with minimum requirements in terms of data. Retraining a model on the
combination of original and new commands, I managed to achieve 40.5% of accuracy on the
new commands with only 10 examples for each of them. This scores goes up to 81.5% of
accuracy with a larger set of 100 examples per new command. An alternative strategy, based
on model adaptation achieved even better scores, with 68.8% and 88.4% of accuracy with 10
and 100 examples respectively, while being faster to train. This high performance is obtained
at the expense of the original categories though, on which the accuracy deteriorated. Those
results are very promising as the methods allow to easily extend an existing S2S model with
minimal resources.
Finally, a full spoken command recognition system (named iCubrec) has been developed
for the iCub platform. The pipeline relies on a voice activity detection (VAD) system to
propose a fully hand-free experience. By segmenting only regions that are likely to contain
commands, the VAD module also allows to reduce greatly the computational cost of the
pipeline. Command candidates are then passed to the deep neural network (DNN)-HMM
command recognition system for transcription. The VoCub dataset has been specifically
gathered to train a DNN-based acoustic model for our task. Through multi-condition training
with the CHiME4 dataset, an accuracy of 94.5% is reached on VoCub test set. A filler model,
complemented by a rejection mechanism based on a confidence score, is finally added to the
system to reject non-command speech in a live demonstration of the system
Out-of-vocabulary spoken term detection
Spoken term detection (STD) is a fundamental task for multimedia information
retrieval. A major challenge faced by an STD system is the serious performance reduction
when detecting out-of-vocabulary (OOV) terms. The difficulties arise not only
from the absence of pronunciations for such terms in the system dictionaries, but from
intrinsic uncertainty in pronunciations, significant diversity in term properties and a
high degree of weakness in acoustic and language modelling.
To tackle the OOV issue, we first applied the joint-multigram model to predict pronunciations
for OOV terms in a stochastic way. Based on this, we propose a stochastic
pronunciation model that considers all possible pronunciations for OOV terms so that
the high pronunciation uncertainty is compensated for.
Furthermore, to deal with the diversity in term properties, we propose a termdependent
discriminative decision strategy, which employs discriminative models to
integrate multiple informative factors and confidence measures into a classification
probability, which gives rise to minimum decision cost.
In addition, to address the weakness in acoustic and language modelling, we propose
a direct posterior confidence measure which replaces the generative models with
a discriminative model, such as a multi-layer perceptron (MLP), to obtain a robust
confidence for OOV term detection.
With these novel techniques, the STD performance on OOV terms was improved
substantially and significantly in our experiments set on meeting speech data
Speech Recognition
Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes
Arbitrary Keyword Spotting in Handwritten Documents
Despite the existence of electronic media in today’s world, a considerable amount of written communications is in paper form such as books, bank cheques, contracts, etc. There is an increasing demand for the automation of information extraction, classification, search, and retrieval of documents. The goal of this research is to develop a complete methodology for the spotting of arbitrary keywords in handwritten document images.
We propose a top-down approach to the spotting of keywords in document images. Our approach is composed of two major steps: segmentation and decision. In the former, we generate the word hypotheses. In the latter, we decide whether a generated word hypothesis is a specific keyword or not. We carry out the decision step through a two-level classification where first, we assign an input image to a keyword or non-keyword class; and then transcribe the image if it is passed as a keyword. By reducing the problem from the image domain to the text domain, we do not only address the search problem in handwritten documents, but also the classification and retrieval, without the need for the transcription of the whole document image.
The main contribution of this thesis is the development of a generalized minimum edit distance for handwritten words, and to prove that this distance is equivalent to an Ergodic Hidden Markov Model (EHMM). To the best of our knowledge, this work is the first to present an exact 2D model for the temporal information in handwriting while satisfying practical constraints.
Some other contributions of this research include: 1) removal of page margins based on corner detection in projection profiles; 2) removal of noise patterns in handwritten images using expectation maximization and fuzzy inference systems; 3) extraction of text lines based on fast Fourier-based steerable filtering; 4) segmentation of characters based on skeletal graphs; and 5) merging of broken characters based on graph partitioning.
Our experiments with a benchmark database of handwritten English documents and a real-world collection of handwritten French documents indicate that, even without any word/document-level training, our results are comparable with two state-of-the-art word spotting systems for English and French documents
Application of automatic speech recognition technologies to singing
The research field of Music Information Retrieval is concerned with the automatic analysis of musical characteristics. One aspect that has not received much attention so far is the automatic analysis of sung lyrics. On the other hand, the field of Automatic Speech Recognition has produced many methods for the automatic analysis of speech, but those have rarely been employed for singing. This thesis analyzes the feasibility of applying various speech recognition methods to singing, and suggests adaptations. In addition, the routes to practical applications for these systems are described. Five tasks are considered: Phoneme recognition, language identification, keyword spotting, lyrics-to-audio alignment, and retrieval of lyrics from sung queries. The main bottleneck in almost all of these tasks lies in the recognition of phonemes from sung audio. Conventional models trained on speech do not perform well when applied to singing. Training models on singing is difficult due to a lack of annotated data. This thesis offers two approaches for generating such data sets. For the first one, speech recordings are made more “song-like”. In the second approach, textual lyrics are automatically aligned to an existing singing data set. In both cases, these new data sets are then used for training new acoustic models, offering considerable improvements over models trained on speech. Building on these improved acoustic models, speech recognition algorithms for the individual tasks were adapted to singing by either improving their robustness to the differing characteristics of singing, or by exploiting the specific features of singing performances. Examples of improving robustness include the use of keyword-filler HMMs for keyword spotting, an i-vector approach for language identification, and a method for alignment and lyrics retrieval that allows highly varying durations. Features of singing are utilized in various ways: In an approach for language identification that is well-suited for long recordings; in a method for keyword spotting based on phoneme durations in singing; and in an algorithm for alignment and retrieval that exploits known phoneme confusions in singing.Das Gebiet des Music Information Retrieval befasst sich mit der automatischen Analyse von musikalischen Charakteristika. Ein Aspekt, der bisher kaum erforscht wurde, ist dabei der gesungene Text. Auf der anderen Seite werden in der automatischen Spracherkennung viele Methoden für die automatische Analyse von Sprache entwickelt, jedoch selten für Gesang. Die vorliegende Arbeit untersucht die Anwendung von Methoden aus der Spracherkennung auf Gesang und beschreibt mögliche Anpassungen. Zudem werden Wege zur praktischen Anwendung dieser Ansätze aufgezeigt. Fünf Themen werden dabei betrachtet: Phonemerkennung, Sprachenidentifikation, Schlagwortsuche, Text-zu-Gesangs-Alignment und Suche von Texten anhand von gesungenen Anfragen. Das größte Hindernis bei fast allen dieser Themen ist die Erkennung von Phonemen aus Gesangsaufnahmen. Herkömmliche, auf Sprache trainierte Modelle, bieten keine guten Ergebnisse für Gesang. Das Trainieren von Modellen auf Gesang ist schwierig, da kaum annotierte Daten verfügbar sind. Diese Arbeit zeigt zwei Ansätze auf, um solche Daten zu generieren. Für den ersten wurden Sprachaufnahmen künstlich gesangsähnlicher gemacht. Für den zweiten wurden Texte automatisch zu einem vorhandenen Gesangsdatensatz zugeordnet. Die neuen Datensätze wurden zum Trainieren neuer Modelle genutzt, welche deutliche Verbesserungen gegenüber sprachbasierten Modellen bieten. Auf diesen verbesserten akustischen Modellen aufbauend wurden Algorithmen aus der Spracherkennung für die verschiedenen Aufgaben angepasst, entweder durch das Verbessern der Robustheit gegenüber Gesangscharakteristika oder durch das Ausnutzen von hilfreichen Besonderheiten von Gesang. Beispiele für die verbesserte Robustheit sind der Einsatz von Keyword-Filler-HMMs für die Schlagwortsuche, ein i-Vector-Ansatz für die Sprachenidentifikation sowie eine Methode für das Alignment und die Textsuche, die stark schwankende Phonemdauern nicht bestraft. Die Besonderheiten von Gesang werden auf verschiedene Weisen genutzt: So z.B. in einem Ansatz für die Sprachenidentifikation, der lange Aufnahmen benötigt; in einer Methode für die Schlagwortsuche, die bekannte Phonemdauern in Gesang mit einbezieht; und in einem Algorithmus für das Alignment und die Textsuche, der bekannte Phonemkonfusionen verwertet
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