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Ana I. García-Moral

dblp:68/2215 · DBLP profile ↗
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4ranked-venue papers
1as first author
0since 2021 · last 2012
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 4 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 2

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Artificial intelligence
1 paper
Speech recognition and synthesis · 75% Learning paradigms · 25%
Computer graphics and multimedia
1 paper
Audio and music processing · 100%

Topics — the 7 heaviest of 7, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing › speech recognition
acoustic modeling
0.112012
Real-Time Robust Automatic Speech Recognition Using Compact Support Vector Machines · IEEE Trans. Speech Audio Process. 2012
Audio and music processing
speech recognition
0.112012
Real-Time Robust Automatic Speech Recognition Using Compact Support Vector Machines · IEEE Trans. Speech Audio Process. 2012
Natural language and speech › Speech recognition and synthesis
acoustic modeling
0.112011
Data Balancing for Efficient Training of Hybrid ANN/HMM Automatic Speech Recognition Systems · IEEE Trans. Speech Audio Process. 2011
Natural language and speech › Speech recognition and synthesis
automatic speech recognition
0.112011
Data Balancing for Efficient Training of Hybrid ANN/HMM Automatic Speech Recognition Systems · IEEE Trans. Speech Audio Process. 2011
Natural language and speech › Speech recognition and synthesis › acoustic modeling
hybrid HMM/ANN
0.112011
Data Balancing for Efficient Training of Hybrid ANN/HMM Automatic Speech Recognition Systems · IEEE Trans. Speech Audio Process. 2011
Machine learning › Learning paradigms › data balancing
training data balancing
0.112011
Data Balancing for Efficient Training of Hybrid ANN/HMM Automatic Speech Recognition Systems · IEEE Trans. Speech Audio Process. 2011
Audio and music processing › speech recognition
hidden markov model
0.012012
Real-Time Robust Automatic Speech Recognition Using Compact Support Vector Machines · IEEE Trans. Speech Audio Process. 2012

Methods — techniques the papers use, named apart from their topics

weighted least squares · 0.1support vector machine · 0.1compact semiparametric model · 0.1posterior probability scaling · 0.1class balancing · 0.1
YearPublicationVenuePosition
2012 Real-Time Robust Automatic Speech Recognition Using Compact Support Vector Machines
abstract
In the last years, support vector machines (SVMs) have shown excellent performance in many applications, especially in the presence of noise. In particular, SVMs offer several advantages over artificial neural networks (ANNs) that have attracted the attention of the speech processing community. Nevertheless, their high computational requirements prevent them from being used in practice in automatic speech recognition (ASR), where ANNs have proven to be successful. The high complexity of SVMs in this context arises from the use of huge speech training databases with millions of samples and highly overlapped classes. This paper suggests the use of a weighted least squares (WLS) training procedure that facilitates the possibility of imposing a compact semiparametric model on the SVM, which results in a dramatic complexity reduction. Such a complexity reduction with respect to conventional SVMs, which is between two and three orders of magnitude, allows the proposed hybrid WLS-SVC/HMM system to perform real-time speech decoding on a connected-digit recognition task (SpeechDat Spanish database). The experimental evaluation of the proposed system shows encouraging performance levels in clean and noisy conditions, although further improvements are required to reach the maturity level of current context-dependent HMM-based recognizers.
Rubén Solera-Ureña, Ana I. García-Moral, Carmen Peláez-Moreno, Manel Martínez-Ramón, Fernando Díaz-de-María
IEEE Trans. Speech Audio Process.2
2011 Data Balancing for Efficient Training of Hybrid ANN/HMM Automatic Speech Recognition Systems
abstract
Hybrid speech recognizers, where the estimation of the emission pdf of the states of hidden Markov models (HMMs), usually carried out using Gaussian mixture models (GMMs), is substituted by artificial neural networks (ANNs) have several advantages over the classical systems. However, to obtain performance improvements, the computational requirements are heavily increased because of the need to train the ANN. Departing from the observation of the remarkable skewness of speech data, this paper proposes sifting out the training set and balancing the amount of samples per class. With this method, the training time has been reduced 18 times while obtaining performances similar to or even better than those with the whole database, especially in noisy environments. However, the application of these reduced sets is not straightforward. To avoid the mismatch between training and testing conditions created by the modification of the distribution of the training data, a proper scaling of the a posteriori probabilities obtained and a resizing of the context window need to be performed as demonstrated in this paper.
Ana I. García-Moral, Rubén Solera-Ureña, Carmen Peláez-Moreno, Fernando Díaz-de-María
IEEE Trans. Speech Audio Process.1
2009 Eliciting a hierarchical structure of human consonant perception task errors using formal concept analysis
Carmen Peláez-Moreno, Ana I. García-Moral, Francisco J. Valverde-Albacete
INTERSPEECH2
2005 Design of a voice-enabled interface for real-time access to stock exchange from a PDA through GPRS
Darío Martín-Iglesias, Yago Pereiro-Estevan, Ana I. García-Moral, Ascensión Gallardo-Antolín, Fernando Díaz-de-María
INTERSPEECH3