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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Wei Zhou Haotian Wu Jingjing Xu Mohammad Zeineldeen Christoph Lüscher Ralf Schlüter Hermann Ney Computer Science Department Human Language Technology and Pattern Recognition RWTH Aachen University Aachen Germany AppTek GmbH Aachen Germany

ASR can be improved by multi-task learning (MTL) with domain enhancing or domain adversarial training, which are two opposite objectives with the aim to increase/decrease domain variance towards domain-aware/agnostic ASR, respectively. In this work, we study how to best apply these two opposite objectives with speaker labels to improve conformer-based ASR. We also propose a novel adaptive gradient reversal layer for stable and effective adversarial training without tuning effort. Detailed analysis and experimental verification are conducted to show the optimal positions in the ASR neural network (NN) to apply speaker enhancing and adversarial training. We also explore their combination for further improvement, achieving the same performance as i-vectors plus adversarial training. Our best speaker-based MTL achieves 7% relative improvement on the Switchboard Hub5’00 set. We also investigate the effect of such speaker-based MTL w.r.t. cleaner dataset and weaker ASR NN.

关键词： Training Artificial neural networks Switches Signal processing Multitasking Acoustics Speech processing

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Influence of text line segmentation in Handwritten Text recognition 13

Influence of text line segmentation in Handwritten Text Reco...

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13th International Conference on Document Analysis and recognition, ICDAR 2015

作者： Romero, Veronica Sanchez, Joan Andreu Bosch, Vicente Depuydt, Katrien De Does, Jesse Pattern Recognition and Human Language Technology Universitat Politècnica de València Camí de Vera S/n València Spain Instituut voor Nederlandse Lexicologie 2300RA Matthias de Vrieshof 2-3 Leiden2311 BZ Netherlands

ISBN: (纸本)9781479918058

Text line segmentation is the process by which text lines in a document image are localized and extracted. It is an important step in off-line Handwritten Text recognition (HTR) given that the input of these systems is the line image of the text to be transcribed. A myriad of solutions to the text line segmentation problem have been proposed in the literature. Although these solutions may differ greatly on what is actually applied to perform the segmentation, they can be classified by the level of precision and detail in the final extracted lines. In this paper we study the influence and real needs of different levels of precision and detail in the segmentation solutions in a real HTR task. We test three technics of text line segmentation whose output range from a simple rectangle for each line to a perfect fitted polygon surrounding the detected lines. Experiments have been carried out with a historical collection and results show that good HTR accuracy can be obtained with simple extraction algorithms. © 2015 IEEE.

关键词： Image segmentation

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On the Relation Between Internal language Model and Sequence Discriminative Training for Neural Transducers

On the Relation Between Internal Language Model and Sequence...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Zijian Yang Wei Zhou Ralf Schlüter Hermann Ney Computer Science Department Human Language Technology and Pattern Recognition RWTH Aachen University Aachen Germany AppTek GmbH Aachen Germany

Internal language model (ILM) subtraction has been widely applied to improve the performance of the RNN-Transducer with external language model (LM) fusion for speech recognition. In this work, we show that sequence discriminative training has a strong correlation with ILM subtraction from both theoretical and empirical points of view. Theoretically, we derive that the global optimum of maximum mutual information (MMI) training shares a similar formula as ILM subtraction. Empirically, we show that ILM subtraction and sequence discriminative training achieve similar effects across a wide range of experiments on Librispeech, including both MMI and minimum Bayes risk (MBR) criteria, as well as neural transducers and LMs of both full and limited context. The benefit of ILM subtraction also becomes much smaller after sequence discriminative training. We also provide an indepth study to show that sequence discriminative training has a minimal effect on the commonly used zero-encoder ILM estimation, but a joint effect on both encoder and prediction + joint network for posterior probability reshaping including both ILM and blank suppression.

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Lattice-Free Sequence Discriminative Training for Phoneme-Based Neural Transducers

Lattice-Free Sequence Discriminative Training for Phoneme-Ba...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

Recently, RNN-Transducers have achieved remarkable results on various automatic speech recognition tasks. However, lattice-free sequence discriminative training methods, which obtain superior performance in hybrid models, are rarely investigated in RNN-Transducers. In this work, we propose three lattice-free training objectives, namely lattice-free maximum mutual information, lattice-free segment-level minimum Bayes risk, and lattice-free minimum Bayes risk, which are used for the final posterior output of the phoneme-based neural transducer with a limited context dependency. Compared to criteria using N-best lists, lattice-free methods eliminate the decoding step for hypotheses generation during training, which leads to more efficient training. Experimental results show that lattice-free methods gain up to 6.5% relative improvement in word error rate compared to a sequence-level cross-entropy trained model. Compared to the N-best-list based minimum Bayes risk objectives, lattice-free methods gain 40% - 70% relative training time speedup with a small degradation in performance.

关键词： Training Degradation Transducers Error analysis Signal processing Decoding Speech processing

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Why does CTC result in peaky behavior?

arXiv

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arXiv 2021年

作者： Zeyer, Albert Schlüter, Ralf Ney, Hermann Human Language Technology and Pattern Recognition Computer Science Department RWTH Aachen University Aachen Germany AppTek GmbH Aachen Germany

The peaky behavior of CTC models is well known experimentally. However, an understanding about why peaky behavior occurs is missing, and whether this is a good property. We provide a formal analysis of the peaky behavior and gradient descent convergence properties of the CTC loss and related training criteria. Our analysis provides a deep understanding why peaky behavior occurs and when it is suboptimal. On a simple example which should be trivial to learn for any model, we prove that a feed-forward neural network trained with CTC from uniform initialization converges towards peaky behavior with a 100% error rate. Our analysis further explains why CTC only works well together with the blank label. We further demonstrate that peaky behavior does not occur on other related losses including a label prior model, and that this improves convergence. © 2021, CC BY-SA.

关键词： Gradient methods

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One decade of statistical machine translation: 1996-2005

One decade of statistical machine translation: 1996-2005

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IEEE Workshop on Automatic Speech recognition and Understanding

作者： H. Ney Human Language Technology and Pattern Recognition Lehrstuhl fur Informatik VI-Computer Science Department RWTH Aachen University Aachen Germany

In the last decade, the statistical approach has found widespread use in machine translation both for written and spoken language and has had a major impact on the translation accuracy. The goal of this paper is to cover the state of the art in statistical machine translation. We would re-visit the underlying principles of the statistical approach to machine translation and summarize the progress that has been made over the last decade

关键词： Natural languages Context modeling Hidden Markov models Speech humans pattern recognition Computer science Transducers Tides Vocabulary

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COMPARISON OF FEEDFORWARD AND RECURRENT NEURAL NETWORK language MODELS

COMPARISON OF FEEDFORWARD AND RECURRENT NEURAL NETWORK LANGU...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： M. Sundermeyer I. Oparin J.-L. Gauvain B. Freiberg R. Schluter H. Ney Human Language Technology and Pattern Recognition Computer Science Department RWTH Aachen University Aachen Germany Spoken Language Processing Group LIMSI CNRS Paris France

ISBN: (纸本)9781479903573

Research on language modeling for speech recognition has increasingly focused on the application of neural networks. Two competing concepts have been developed: On the one hand, feedforward neural networks representing an ngram approach, on the other hand recurrent neural networks that may learn context dependencies spanning more than a fixed number of predecessor words. To the best of our knowledge, no comparison has been carried out between feedforward and state-of-the-art recurrent networks when applied to speech recognition. This paper analyzes this aspect in detail on a well-tuned French speech recognition task. In addition, we propose a simple and efficient method to normalize language model probabilities across different vocabularies, and we show how to speed up training of recurrent neural networks by parallelization.

关键词： Automatic speech recognition feedforward neural networks recurrent neural networks Speech recognition recurrent neural nets feedforward neural nets FEEDFORWARD modelling languages Neural network parallelization

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THE RWTH ENGLISH LECTURE recognition SYSTEM

THE RWTH ENGLISH LECTURE RECOGNITION SYSTEM

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Simon Wiesler Kazuki Irie Zoltan Tuske Ralf Schluter Hermann Ney Human Language Technology and Pattern Recognition Computer Science Department RWTH Aachen University Aachen Germany Ecole Centrale Paris Paris France

ISBN: (纸本)9781479928941

In this paper, we describe the RWTH speech recognition system for English lectures developed within the Translectures project. A difficulty in the development of an English lectures recognition system, is the high ratio of non-native speakers. We address this problem by using very effective deep bottleneck features trained on multilingual data. The acoustic model is trained on large amounts of data from different domains and with different dialects. Large improvements are obtained from unsupervised acoustic adaptation. Another challenge is the frequent use of technical terms and the wide range of topics. In our recognition system, slides, which are attached to most lectures, are used for improving lexical coverage and language model adaptation.

关键词： Lectures (teaching method) recognition system modelling languages speech recognition system English English language Acoustic models

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Efficient Utilization of Large Pre-Trained Models for Low Resource ASR

Efficient Utilization of Large Pre-Trained Models for Low Re...

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Acoustics, Speech, and Signal Processing Workshops (ICASSPW), IEEE International Conference on

作者： Peter Vieting Christoph Lüscher Julian Dierkes Ralf Schlüter Hermann Ney Computer Science Department Human Language Technology and Pattern Recognition Group RWTH Aachen University Aachen Germany AppTek GmbH Aachen Germany

Unsupervised representation learning has recently helped automatic speech recognition (ASR) to tackle tasks with limited labeled data. Following this, hardware limitations and applications give rise to the question how to take advantage of large pre-trained models efficiently and reduce their complexity. In this work, we study a challenging low resource conversational telephony speech corpus from the medical domain in Vietnamese and German. We show the benefits of using unsupervised techniques beyond simple fine-tuning of large pre-trained models, discuss how to adapt them to a practical telephony task including bandwidth transfer and investigate different data conditions for pre-training and fine-tuning. We outperform the project baselines by 22% relative using pre-training techniques. Further gains of 29% can be achieved by refinements of architecture and training and 6% by adding 0.8 h of in-domain adaptation data.

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Acoustic feature combination for robust speech recognition

Acoustic feature combination for robust speech recognition

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： A. Zolnay R. Schluter H. Ney Human Language Technology and Pattern Recognition Lehrstuhl für Informatik VI Computer Science Department RWTH Aachen University Aachen Germany

In this paper, we consider the use of multiple acoustic features of the speech signal for robust speech recognition. We investigate the combination of various auditory based (mel frequency cepstrum coefficients, perceptual linear prediction, etc.) and articulatory based (voicedness) features. Features are combined by linear discriminant analysis and log-linear model combination based techniques. We describe the two feature combination techniques and compare the experimental results. Experiments performed on the large-vocabulary task VerbMobil II (German conversational speech) show that the accuracy of automatic speech recognition systems can be improved by the combination of different acoustic features.

关键词： Robustness Speech recognition Mel frequency cepstral coefficient Linear discriminant analysis Cepstrum Feature extraction Automatic speech recognition Autocorrelation Cepstral analysis Concatenated codes

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