Demonstration venue · read-only. Every page can be browsed; the buttons that would change it are switched off. Create an account to run TaxoReview on your own data. A simulation from public publication records: the people listed were not invited and did not agree to review.
? Tutorial for this page

Lakshmi Babu Saheer

dblp:266/3448 · DBLP profile ↗
← Back
7ranked-venue papers
4as first author
0since 2021 · last 2013
0000-0002-5263-596XORCID · corroborated

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

Graphics, computer vision, multimedia, augmented reality and games · 4 · 2 first-authorArtificial intelligence and machine learning · 3 · 1 first-authorHuman-computer interaction and ubiquitous computing · 1 · 1 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.

Computer graphics and multimedia
1 paper
Audio and music processing · 100%

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

TopicWeightPapersLastEvidence papers
Audio and music processing
speech synthesis
0.112012
Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis · IEEE Trans. Speech Audio Process. 2012
Audio and music processing › speech synthesis
statistical parametric speech synthesis
0.112012
Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis · IEEE Trans. Speech Audio Process. 2012
Audio and music processing › speech recognition › speaker normalization
vocal tract length normalization
0.112012
Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis · IEEE Trans. Speech Audio Process. 2012
Audio and music processing › speech recognition
speaker adaptation
0.012012
Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis · IEEE Trans. Speech Audio Process. 2012

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

jacobian normalization · 0.1expectation-maximization · 0.1
YearPublicationVenuePosition
2013 Automatic Staging of Audio with Emotions
abstract
Current day text-to-speech technologies are mature enough to be acceptable in quality for the users. There is still a large gap between a synthesised speech and a real human speech due to lack of expressions and emotions. Geneemo is a technology for automatic addition of emotions and expressions to any audio. The process of staging the text is to dramatize it. The text is enriched and transformed into a performance. Similarly, "staging the audio" refers to extending text dramatisation to audio by enriching emotionally neutral audio content into a natural human speech with real expressions. The audio can be generated by any text-to-speech technology. The aim of the project is to make human computer interactions as natural as possible with expressive speech. This also opens up a portfolio of applications replacing real human voices.
Lakshmi Babu Saheer, Milos Cernak
ACII1
2013 Personalising speech-to-speech translation: Unsupervised cross-lingual speaker adaptation for HMM-based speech synthesis
John Dines, Lakshmi Babu Saheer, Matthew Gibson, William J. Byrne, Keiichiro Oura, Keiichi Tokuda, Junichi Yamagishi, Simon King 0001, Mirjam Wester, Teemu Hirsimäki, Reima Karhila, Mikko Kurimo
Comput. Speech Lang.3
2012 Combining vocal tract length normalization with hierarchial linear transformations
abstract
Recent research has demonstrated the effectiveness of vocal tract length normalization (VTLN) as a rapid adaptation technique for statistical parametric speech synthesis. VTLN produces speech with naturalness preferable to that of MLLR-based adaptation techniques, being much closer in quality to that generated by the original average voice model. However with only a single parameter, VTLN captures very few speaker specific characteristics when compared to linear transform based adaptation techniques. This paper proposes that the merits of VTLN can be combined with those of linear transform based adaptation in a hierarchial Bayesian framework, where VTLN is used as the prior information. A novel technique for propagating the gender information from the VTLN prior through constrained structural maximum a posteriori linear regression (CSMAPLR) adaptation is presented. Experiments show that the resulting transformation has improved speech quality with better naturalness, intelligibility and improved speaker similarity.
Lakshmi Babu Saheer, Junichi Yamagishi, Philip N. Garner, John Dines
ICASSP1
2012 Vocal Tract Length Normalization for Statistical Parametric Speech Synthesis
abstract
Vocal tract length normalization (VTLN) has been successfully used in automatic speech recognition for improved performance. The same technique can be implemented in statistical parametric speech synthesis for rapid speaker adaptation during synthesis. This paper presents an efficient implementation of VTLN using expectation maximization and addresses the key challenges faced in implementing VTLN for synthesis. Jacobian normalization, high-dimensionality features and truncation of the transformation matrix are a few challenges presented with the appropriate solutions. Detailed evaluations are performed to estimate the most suitable technique for using VTLN in speech synthesis. Evaluating VTLN in the framework of speech synthesis is also not an easy task since the technique does not work equally well for all speakers. Speakers have been selected based on different objective and subjective criteria to demonstrate the difference between systems. The best method for implementing VTLN is confirmed to be use of the lower order features for estimating warping factors.
Lakshmi Babu Saheer, John Dines, Philip N. Garner
IEEE Trans. Speech Audio Process.1
2010 A comparison of supervised and unsupervised cross-lingual speaker adaptation approaches for HMM-based speech synthesis
abstract
The EMIME project aims to build a personalized speech-to-speech translator, such that spoken input of a user in one language is used to produce spoken output that still sounds like the user's voice however in another language. This distinctiveness makes unsupervised cross-lingual speaker adaptation one key to the project's success. So far, research has been conducted into unsupervised and cross-lingual cases separately by means of decision tree marginalization and HMM state mapping respectively. In this paper we combine the two techniques to perform unsupervised cross-lingual speaker adaptation. The performance of eight speaker adaptation systems (supervised vs. unsupervised, intra-lingual vs. cross-lingual) is compared using objective and subjective evaluations. Experimental results show the performance of unsupervised cross-lingual speaker adaptation is comparable to that of the supervised case in terms of spectrum adaptation in the EMIME scenario, even though automatically obtained transcriptions have a very high phoneme error rate.
John Dines, Lakshmi Babu Saheer
ICASSP3
2010 VTLN adaptation for statistical speech synthesis
abstract
The advent of statistical speech synthesis has enabled the unification of the basic techniques used in speech synthesis and recognition. Adaptation techniques that have been successfully used in recognition systems can now be applied to synthesis systems to improve the quality of the synthesized speech. The application of vocal tract length normalization (VTLN) for synthesis is explored in this paper. VTLN based adaptation requires estimation of a single warping factor, which can be accurately estimated from very little adaptation data and gives additive improvements over CMLLR adaptation. The challenge of estimating accurate warping factors using higher order features is solved by initializing warping factor estimation with the values calculated from lower order features.
Lakshmi Babu Saheer, Philip N. Garner, John Dines
ICASSP1
2009 Speech recognition with speech synthesis models by marginalising over decision tree leaves
abstract
There has been increasing interest in the use of unsupervised adaptation for the personalisation of text-to-speech (TTS) voices, particularly in the context of speech-to-speech translation. This requires that we are able to generate adaptation transforms from the output of an automatic speech recognition (ASR) system. An approach that utilises unified ASR and TTS models would seem to offer an ideal mechanism for the application of unsupervised adaptation to TTS since transforms could be shared between ASR and TTS. Such unified models should use a common set of parameters. A major barrier to such parameter sharing is the use of differing contexts in ASR and TTS. In this paper we propose a simple approach that generates ASR models from a trained set of TTS models by marginalising over the TTS contexts that are not used by ASR. We present preliminary results of our proposed method on a large vocabulary speech recognition task and provide insights into future directions of this work. Index Terms: speech synthesis, speech recognition, decision trees, unified models
John Dines, Lakshmi Babu Saheer
INTERSPEECH2