Context-Dependent Acoustic Modeling without Explicit Phone Clustering
- URL: http://arxiv.org/abs/2005.07578v2
- Date: Wed, 7 Apr 2021 12:32:37 GMT
- Title: Context-Dependent Acoustic Modeling without Explicit Phone Clustering
- Authors: Tina Raissi, Eugen Beck, Ralf Schl\"uter, Hermann Ney
- Abstract summary: Phoneme-based acoustic modeling of large vocabulary automatic speech recognition takes advantage of phoneme context.
In this work, we address a direct phonetic context modeling for the hybrid deep neural network (DNN)/HMM.
By performing different decompositions of the joint probability of the center phoneme state and its left and right contexts, we obtain a factorized network consisting of different components.
- Score: 45.07737874541304
- License: http://arxiv.org/licenses/nonexclusive-distrib/1.0/
- Abstract: Phoneme-based acoustic modeling of large vocabulary automatic speech
recognition takes advantage of phoneme context. The large number of
context-dependent (CD) phonemes and their highly varying statistics require
tying or smoothing to enable robust training. Usually, classification and
regression trees are used for phonetic clustering, which is standard in hidden
Markov model (HMM)-based systems. However, this solution introduces a secondary
training objective and does not allow for end-to-end training. In this work, we
address a direct phonetic context modeling for the hybrid deep neural network
(DNN)/HMM, that does not build on any phone clustering algorithm for the
determination of the HMM state inventory. By performing different
decompositions of the joint probability of the center phoneme state and its
left and right contexts, we obtain a factorized network consisting of different
components, trained jointly. Moreover, the representation of the phonetic
context for the network relies on phoneme embeddings. The recognition accuracy
of our proposed models on the Switchboard task is comparable and outperforms
slightly the hybrid model using the standard state-tying decision trees.
Related papers
- The OCON model: an old but green solution for distributable supervised classification for acoustic monitoring in smart cities [0.28675177318965045]
This paper focuses on vowel phonemes classification and speakers recognition for the Automatic Speech Recognition domain.
For our case-study, the ASR model runs on a proprietary sensing and lightning system, exploited to monitor acoustic and air pollution on urban streets.
We formalize combinations of pseudo-Neural Architecture Search and Hyper-s Tuning experiments, using an informed grid-search methodology, to achieve classification accuracy comparable to nowadays most complex architectures.
arXiv Detail & Related papers (2024-10-05T09:47:54Z) - Improved Contextual Recognition In Automatic Speech Recognition Systems
By Semantic Lattice Rescoring [4.819085609772069]
We propose a novel approach for enhancing contextual recognition within ASR systems via semantic lattice processing.
Our solution consists of using Hidden Markov Models and Gaussian Mixture Models (HMM-GMM) along with Deep Neural Networks (DNN) models for better accuracy.
We demonstrate the effectiveness of our proposed framework on the LibriSpeech dataset with empirical analyses.
arXiv Detail & Related papers (2023-10-14T23:16:05Z) - Continual Learning for On-Device Speech Recognition using Disentangled
Conformers [54.32320258055716]
We introduce a continual learning benchmark for speaker-specific domain adaptation derived from LibriVox audiobooks.
We propose a novel compute-efficient continual learning algorithm called DisentangledCL.
Our experiments show that the DisConformer models significantly outperform baselines on general ASR.
arXiv Detail & Related papers (2022-12-02T18:58:51Z) - Bridging Speech and Textual Pre-trained Models with Unsupervised ASR [70.61449720963235]
This work proposes a simple yet efficient unsupervised paradigm that connects speech and textual pre-trained models.
We show that unsupervised automatic speech recognition (ASR) can improve the representations from speech self-supervised models.
Notably, on spoken question answering, we reach the state-of-the-art result over the challenging NMSQA benchmark.
arXiv Detail & Related papers (2022-11-06T04:50:37Z) - A Correspondence Variational Autoencoder for Unsupervised Acoustic Word
Embeddings [50.524054820564395]
We propose a new unsupervised model for mapping a variable-duration speech segment to a fixed-dimensional representation.
The resulting acoustic word embeddings can form the basis of search, discovery, and indexing systems for low- and zero-resource languages.
arXiv Detail & Related papers (2020-12-03T19:24:42Z) - Phoneme Based Neural Transducer for Large Vocabulary Speech Recognition [41.92991390542083]
We present a simple, novel and competitive approach for phoneme-based neural transducer modeling.
A phonetic context size of one is shown to be sufficient for the best performance.
The overall performance of our best model is comparable to state-of-the-art (SOTA) results for the TED-LIUM Release 2 and Switchboard corpora.
arXiv Detail & Related papers (2020-10-30T16:53:29Z) - Any-to-Many Voice Conversion with Location-Relative Sequence-to-Sequence
Modeling [61.351967629600594]
This paper proposes an any-to-many location-relative, sequence-to-sequence (seq2seq), non-parallel voice conversion approach.
In this approach, we combine a bottle-neck feature extractor (BNE) with a seq2seq synthesis module.
Objective and subjective evaluations show that the proposed any-to-many approach has superior voice conversion performance in terms of both naturalness and speaker similarity.
arXiv Detail & Related papers (2020-09-06T13:01:06Z) - AutoSpeech: Neural Architecture Search for Speaker Recognition [108.69505815793028]
We propose the first neural architecture search approach approach for the speaker recognition tasks, named as AutoSpeech.
Our algorithm first identifies the optimal operation combination in a neural cell and then derives a CNN model by stacking the neural cell for multiple times.
Results demonstrate that the derived CNN architectures significantly outperform current speaker recognition systems based on VGG-M, ResNet-18, and ResNet-34 back-bones, while enjoying lower model complexity.
arXiv Detail & Related papers (2020-05-07T02:53:47Z) - Statistical Context-Dependent Units Boundary Correction for Corpus-based
Unit-Selection Text-to-Speech [1.4337588659482519]
We present an innovative technique for speaker adaptation in order to improve the accuracy of segmentation with application to unit-selection Text-To-Speech (TTS) systems.
Unlike conventional techniques for speaker adaptation, we aim to use only context dependent characteristics extrapolated with linguistic analysis techniques.
arXiv Detail & Related papers (2020-03-05T12:42:13Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.