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Ana Ramírez López

dblp:172/8153 · DBLP profile ↗
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5ranked-venue papers
3as first author
0since 2021 · last 2017
—ORCID · none

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

Artificial intelligence and machine learning · 3 · 1 first-authorGraphics, computer vision, multimedia, augmented reality and games · 3 · 3 first-author

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 · 100%

Topics — the 4 heaviest of 4, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis
acoustic model adaptation
0.212015
Bounded Conditional Mean Imputation with Observation Uncertainties and Acoustic Model Adaptation · IEEE ACM Trans. Audio Speech Lang. Process. 2015
Natural language and speech › Speech recognition and synthesis › automatic speech recognition › robust speech recognition
noise compensation
0.212015
Bounded Conditional Mean Imputation with Observation Uncertainties and Acoustic Model Adaptation · IEEE ACM Trans. Audio Speech Lang. Process. 2015
Natural language and speech › Speech recognition and synthesis
observation uncertainty
0.212015
Bounded Conditional Mean Imputation with Observation Uncertainties and Acoustic Model Adaptation · IEEE ACM Trans. Audio Speech Lang. Process. 2015
Natural language and speech › Speech recognition and synthesis › automatic speech recognition
robust speech recognition
0.212015
Bounded Conditional Mean Imputation with Observation Uncertainties and Acoustic Model Adaptation · IEEE ACM Trans. Audio Speech Lang. Process. 2015

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

posterior distribution estimation · 0.2missing data imputation · 0.2
YearPublicationVenuePosition
2017 Normal-to-shouted speech spectral mapping for speaker recognition under vocal effort mismatch
abstract
Speaker recognition performance degrades substantially in case of vocal effort mismatch (e.g. shouted vs. normal speech) between test and enrollment utterances. Such a mismatch is often encountered, for example, in forensic speaker recognition. This paper introduces a novel spectral mapping method which, when employed jointly with a statistical mapping technique, converts the Mel-frequency band energies of normal speech towards their counterparts in shouted speech. The aim is to obtain more robust performance in speaker recognition by tackling vocal effort mismatch between enrollment and test utterances. The processing is performed on the speech signal before feature extraction. The proposed approach was evaluated by testing the performance of a state-of-the-art i-vector-based speaker recognition system with and without applying the spectral mapping processing to the enrollment data. The results show that pre-processing with the proposed approach results in considerable improvement in correct identification rates.
Ana Ramírez López, Rahim Saeidi, Lauri Juvela, Paavo Alku
ICASSP1
2017 Speaking Style Conversion from Normal to Lombard Speech Using a Glottal Vocoder and Bayesian GMMs
abstract
Speaking style conversion is the technology of converting natural speech signals from one style to another. In this study, we focus on normal-to-Lombard conversion. This can be used, for example, to enhance the intelligibility of speech in noisy environments. We propose a parametric approach that uses a vocoder to extract speech features. These features are mapped using Bayesian GMMs from utterances spoken in normal style to the corresponding features of Lombard speech. Finally, the mapped features are converted to a Lombard speech waveform with the vocoder. Two vocoders were compared in the proposed normal-to-Lombard conversion: a recently developed glottal vocoder that decomposes speech into glottal flow excitation and vocal tract, and the widely used STRAIGHT vocoder. The conversion quality was evaluated in two subjective listening tests measuring subjective similarity and naturalness. The similarity test results show that the system is able to convert normal speech into Lombard speech for the two vocoders. However, the subjective naturalness of the converted Lombard speech was clearly better using the glottal vocoder in comparison to STRAIGHT.
Ana Ramírez López, Shreyas Seshadri, Lauri Juvela, Okko Johannes Räsänen, Paavo Alku
INTERSPEECH1
2016 Comparing human and automatic speech recognition in a perceptual restoration experiment
Ulpu Remes, Ana Ramírez López, Lauri Juvela, Kalle J. Palomäki, Guy J. Brown, Paavo Alku, Mikko Kurimo
Comput. Speech Lang.2
2015 Designing multichannel source separation based on single-channel source separation
abstract
In this paper, an extension of independent vector analysis (IVA), model-based IVA, is proposed for multichannel source separation. For obtaining better source models, we introduce a single-channel source separation method, and utilize the outputs as source variances in time-frequency-variant Gaussian source model. The demixing matrices are estimated in the same way as a state-of-the-art IVA method, auxiliary-function-based IVA (AuxIVA). Experimental evaluations show that the proposed approach is effective and improves the source separation performance of IVA. In addition, several post-filters aiming to realize multichannel Wiener filter (MWF) are investigated. This setup proves to further increase the performance of IVA. The presented method shows a potential to provide a general way to improve the separation performance from single-channel source separation to multichannel source separation.
Ana Ramírez López, Nobutaka Ono, Ulpu Remes, Kalle J. Palomäki, Mikko Kurimo
ICASSP1
2015 Bounded Conditional Mean Imputation with Observation Uncertainties and Acoustic Model Adaptation
abstract
Automatic speech recognition systems use noise compensation and acoustic model adaptation to increase robustness towards speaker and environmental variation. The current work focuses on noise compensation with bounded conditional mean imputation (BCMI). BCMI approaches are missing-data methods which operate on the assumption that noise-corrupted observations can be divided into reliable and unreliable components. BCMI methods substitute the unreliable components with a clean speech posterior distribution. The posterior means can be used as clean speech estimates and the posterior variances can be introduced in acoustic model likelihood calculation as observation uncertainties. In addition, we propose in the current work that similar uncertainties are introduced in acoustic model adaptation. Evaluation with speech data recorded in diverse public and car environments indicates that the proposed uncertainties improve adaptation performance. When uncertainties were used in acoustic model likelihood calculation and adaptation, the proposed imputation and adaptation system introduced 15%-84% relative error reductions to an uncompensated baseline system performance.
Ulpu Remes, Ana Ramírez López, Kalle J. Palomäki, Mikko Kurimo
IEEE ACM Trans. Audio Speech Lang. Process.2