16 research outputs found
Factorization of Discriminatively Trained i-vector Extractor for Speaker Recognition
In this work, we continue in our research on i-vector extractor for speaker
verification (SV) and we optimize its architecture for fast and effective
discriminative training. We were motivated by computational and memory
requirements caused by the large number of parameters of the original
generative i-vector model. Our aim is to preserve the power of the original
generative model, and at the same time focus the model towards extraction of
speaker-related information. We show that it is possible to represent a
standard generative i-vector extractor by a model with significantly less
parameters and obtain similar performance on SV tasks. We can further refine
this compact model by discriminative training and obtain i-vectors that lead to
better performance on various SV benchmarks representing different acoustic
domains.Comment: Submitted to Interspeech 2019, Graz, Austria. arXiv admin note:
substantial text overlap with arXiv:1810.1318
BUT text-dependent speaker verification system for SdSV challenge
In this paper, we present the winning BUT submission for the text-dependent task of the SdSV challenge 2020. Given the large amount of training data available in this challenge, we explore successful techniques from text-independent systems in the text-dependent scenario. In particular, we trained x-vector extractors on both in-domain and out-of-domain datasets and combine them with i-vectors trained on concatenated MFCCs and bottleneck features, which have proven effective for the text-dependent scenario. Moreover, we proposed the use of phrase-dependent PLDA backend for scoring and its combination with a simple phrase recognizer, which brings up to 63% relative improvement on our development set with respect to using standard PLDA. Finally, we combine our different i-vector and x-vector based systems using a simple linear logistic regression score level fusion, which provides 28% relative improvement on the evaluation set with respect to our best single systemThe work was supported by Czech Ministry of Interior projects Nos. VI20152020025 “DRAPAK” and VI20192022169 “AI
v TiV”, Czech National Science Foundation (GACR) project “NEUREM3” No. 19-26934X, European Union’s Marie
Sklodowska-Curie grant agreement No. 843627, European Union’s Horizon 2020 project no. 833635 - ROXANNE
and by Czech Ministry of Education, Youth and Sports from the National Programme of Sustainability (NPU II) project
“IT4Innovations excellence in science” - LQ1602 and project no. LTAIN19087 “Multi-linguality in speech technologies
The Kaldi Speech Recognition Toolkit
We describe the design of Kaldi, a free, open-source toolkit for speech recognition research. Kaldi provides a speech recognition system based on finite-state automata (using the freely available OpenFst), together with detailed documentation and a comprehensive set of scripts for building complete recognition systems. Kaldi is written is C++, and the core library supports modeling of arbitrary phonetic-context sizes, acoustic modeling with subspace Gaussian mixture models (SGMM) as well as standard Gaussian mixture models, together with all commonly used linear and affine transforms. Kaldi is released under the Apache License v2.0, which is highly nonrestrictive, making it suitable for a wide community of users
Approaches to automatic lexicon learning with limited training examples
Preparation of a lexicon for speech recognition systems can be a significant effort in languages where the written form is not exactly phonetic. On the other hand, in languages where the written form is quite phonetic, some common words are often mispronounced. In this paper, we use a combination of lexicon learning techniques to explore whether a lexicon can be learned when only a small lexicon is available for boot-strapping. We discover that for a phonetic language such as Spanish, it is possible to do that better than what is possible from generic rules or hand-crafted pronunciations. For a more complex language such as English, we find that it is still possible but with some loss of accuracy
SUBSPACE GAUSSIAN MIXTURE MODELS FOR SPEECH RECOGNITION
We describe an acoustic modeling approach in which all phonetic states share a common Gaussian Mixture Model structure, and the means and mixture weights vary in a subspace of the total parameter space. We call this a Subspace Gaussian Mixture Model (SGMM). Globally shared parameters define the subspace. This style of acoustic model allows for a much more compact representation and gives better results than a conventional modeling approach, particularly with smaller amounts of training data