Daniel Korzekwa
Current Organizations / Institutes
The organizations/institutes from the scholar's latest article
Previous Organizations / Institutes
The organizations / institutes from the scholar's other articles in the past ten years
Research Subjects
Daniel Korzekwa has a total of 27 co-authors. Here are the top co-authors by number of shared publications:
Bozena Kostek
3 shared publicationsBartosz Putrycz
University of Oxford
2 shared publicationsRoberto Barra-Chicote
2 shared publicationsThomas Drugman
2 shared publicationsThomas Merritt
Jagiellonian University POLAND
2 shared publications- Abdelhamid Ezzerg
Amazon Alexa AI
1 shared publication
Latest Publications
- Open Access
Remap, warp and attend: Non-parallel many-to-many accent conversion with Normalizing Flows
Regional accents of the same language affect not only how words are pronounced (i.e., phonetic content), but also impact prosodic aspects of speech such as speaking rate and intonation. This paper investigates a novel flow-based approach to accent conversion using normalizing flows. The proposed approach revolves around three steps: remapping the phonetic conditioning, to better match the target accent, warping the duration of the converted speech, to better suit the target phonemes, and an attention mechanism that implicitly aligns source and target speech sequences. The proposed remap-warp-attend system enables adaptation of both phonetic and prosodic aspects of speech while allowing for source and converted speech signals to be of different lengths. Objective and subjective evaluations show that the proposed approach significantly outperforms a competitive CopyCat baseline model in terms of similarity to the target accent, naturalness and intelligibility.
Open AccessRemap, warp and attend: Non-parallel many-to-many accent conversion with Normalizing Flows
Regional accents of the same language affect not only how words are pronounced (i.e., phonetic content), but also impact prosodic aspects of speech such as speaking rate and intonation. This paper investigates a novel flow-based approach to accent conversion using normalizing flows. The proposed approach revolves around three steps: remapping the phonetic conditioning, to better match the target accent, warping the duration of the converted speech, to better suit the target phonemes, and an attention mechanism that implicitly aligns source and target speech sequences. The proposed remap-warp-attend system enables adaptation of both phonetic and prosodic aspects of speech while allowing for source and converted speech signals to be of different lengths. Objective and subjective evaluations show that the proposed approach significantly outperforms a competitive CopyCat baseline model in terms of similarity to the target accent, naturalness and intelligibility.
- Open Access
Automated detection of pronunciation errors in non-native English speech employing deep learning
Despite significant advances in recent years, the existing Computer-Assisted Pronunciation Training (CAPT) methods detect pronunciation errors with a relatively low accuracy (precision of 60% at 40%-80% recall). This Ph.D. work proposes novel deep learning methods for detecting pronunciation errors in non-native (L2) English speech, outperforming the state-of-the-art method in AUC metric (Area under the Curve) by 41%, i.e., from 0.528 to 0.749. One of the problems with existing CAPT methods is the low availability of annotated mispronounced speech needed for reliable training of pronunciation error detection models. Therefore, the detection of pronunciation errors is reformulated to the task of generating synthetic mispronounced speech. Intuitively, if we could mimic mispronounced speech and produce any amount of training data, detecting pronunciation errors would be more effective. Furthermore, to eliminate the need to align canonical and recognized phonemes, a novel end-to-end multi-task technique to directly detect pronunciation errors was proposed. The pronunciation error detection models have been used at Amazon to automatically detect pronunciation errors in synthetic speech to accelerate the research into new speech synthesis methods. It was demonstrated that the proposed deep learning methods are applicable in the tasks of detecting and reconstructing dysarthric speech.
Open AccessAutomated detection of pronunciation errors in non-native English speech employing deep learning
Despite significant advances in recent years, the existing Computer-Assisted Pronunciation Training (CAPT) methods detect pronunciation errors with a relatively low accuracy (precision of 60% at 40%-80% recall). This Ph.D. work proposes novel deep learning methods for detecting pronunciation errors in non-native (L2) English speech, outperforming the state-of-the-art method in AUC metric (Area under the Curve) by 41%, i.e., from 0.528 to 0.749. One of the problems with existing CAPT methods is the low availability of annotated mispronounced speech needed for reliable training of pronunciation error detection models. Therefore, the detection of pronunciation errors is reformulated to the task of generating synthetic mispronounced speech. Intuitively, if we could mimic mispronounced speech and produce any amount of training data, detecting pronunciation errors would be more effective. Furthermore, to eliminate the need to align canonical and recognized phonemes, a novel end-to-end multi-task technique to directly detect pronunciation errors was proposed. The pronunciation error detection models have been used at Amazon to automatically detect pronunciation errors in synthetic speech to accelerate the research into new speech synthesis methods. It was demonstrated that the proposed deep learning methods are applicable in the tasks of detecting and reconstructing dysarthric speech.
- Open Access
- 22 June 2022
- Published by Elsevier in Speech Communication
- Vol. 142, 22-33
Computer-assisted pronunciation training—Speech synthesis is almost all you need
- 22 June 2022
- Published by Elsevier in Speech Communication
- Vol. 142, 22-33
Open AccessComputer-assisted pronunciation training—Speech synthesis is almost all you need
- Open Access
Universal Neural Vocoding with Parallel WaveNet
We present a universal neural vocoder based on Parallel WaveNet, with an additional conditioning network called Audio Encoder. Our universal vocoder offers real-time high-quality speech synthesis on a wide range of use cases. We tested it on 43 internal speakers of diverse age and gender, speaking 20 languages in 17 unique styles, of which 7 voices and 5 styles were not exposed during training. We show that the proposed universal vocoder significantly outperforms speaker-dependent vocoders overall. We also show that the proposed vocoder outperforms several existing neural vocoder architectures in terms of naturalness and universality. These findings are consistent when we further test on more than 300 open-source voices.
Open AccessUniversal Neural Vocoding with Parallel WaveNet
We present a universal neural vocoder based on Parallel WaveNet, with an additional conditioning network called Audio Encoder. Our universal vocoder offers real-time high-quality speech synthesis on a wide range of use cases. We tested it on 43 internal speakers of diverse age and gender, speaking 20 languages in 17 unique styles, of which 7 voices and 5 styles were not exposed during training. We show that the proposed universal vocoder significantly outperforms speaker-dependent vocoders overall. We also show that the proposed vocoder outperforms several existing neural vocoder architectures in terms of naturalness and universality. These findings are consistent when we further test on more than 300 open-source voices.
- 1 October 2019
- Published by Acoustical Society of America (ASA) in The Journal of the Acoustical Society of America
- Vol. 146 (4_Suppleme) , 2956-2957
Deep learning model for automated assessment of lexical stress of non-native english speakers
In this paper, we present a novel system to practice lexical stress in L2 English learning with Amazon Alexa home assistant. The language learning for non-native English speakers mostly focuses on practicing correct grammar, extending language vocabulary, and improving pronunciation. The system proposed enables a person to practice lexical stress skills at home by having conversations with Alexa assistant. The system assesses student's abilities to enunciate words with a correct lexical stress and automatically selects the next words to practice. After a series of exercises, the system informs the student on the improvement. The main scientific contribution of the work presented is a deep learning model for automated assessment of lexical stress of non-native English speakers. The model is based on a transfer learning technique. First, we train the model to predict the location of a lexical stress on a syllable level using a large corpus of native English speech. Then, we tune the model with a limited amount of a non-native speech. A corpus of non-native English speech obtained from Polish speakers is incorporated into the training and testing of the model. It is shown that the system enables to create a vocabulary for a particular speaker interactively.
- 1 October 2019
- Published by Acoustical Society of America (ASA) in The Journal of the Acoustical Society of America
- Vol. 146 (4_Suppleme) , 2956-2957
Deep learning model for automated assessment of lexical stress of non-native english speakers
In this paper, we present a novel system to practice lexical stress in L2 English learning with Amazon Alexa home assistant. The language learning for non-native English speakers mostly focuses on practicing correct grammar, extending language vocabulary, and improving pronunciation. The system proposed enables a person to practice lexical stress skills at home by having conversations with Alexa assistant. The system assesses student's abilities to enunciate words with a correct lexical stress and automatically selects the next words to practice. After a series of exercises, the system informs the student on the improvement. The main scientific contribution of the work presented is a deep learning model for automated assessment of lexical stress of non-native English speakers. The model is based on a transfer learning technique. First, we train the model to predict the location of a lexical stress on a syllable level using a large corpus of native English speech. Then, we tune the model with a limited amount of a non-native speech. A corpus of non-native English speech obtained from Polish speakers is incorporated into the training and testing of the model. It is shown that the system enables to create a vocabulary for a particular speaker interactively.
- 15 September 2019
- Published by International Speech Communication Association
- p. 3890-3894
Interpretable Deep Learning Model for the Detection and Reconstruction of Dysarthric Speech
- 15 September 2019
- Published by International Speech Communication Association
- p. 3890-3894
Interpretable Deep Learning Model for the Detection and Reconstruction of Dysarthric Speech
- 1 December 2018
- Published by Institute of Electrical and Electronics Engineers (IEEE)
- p. 325-331
Comprehensive Evaluation of Statistical Speech Waveform Synthesis
Statistical TTS systems that directly predict the speech waveform have recently reported improvements in synthesis quality. This investigation evaluates Amazon's statistical speech waveform synthesis (SSWS) system. An in-depth evaluation of SSWS is conducted across a number of domains to better understand the consistency in quality. The results of this evaluation are validated by repeating the procedure on a separate group of testers. Finally, an analysis of the nature of speech errors of SSWS compared to hybrid unit selection synthesis is conducted to identify the strengths and weaknesses of SSWS. Having a deeper insight into SSWS allows us to better define the focus of future work to improve this new technology.
- 1 December 2018
- Published by Institute of Electrical and Electronics Engineers (IEEE)
- p. 325-331
Comprehensive Evaluation of Statistical Speech Waveform Synthesis
Statistical TTS systems that directly predict the speech waveform have recently reported improvements in synthesis quality. This investigation evaluates Amazon's statistical speech waveform synthesis (SSWS) system. An in-depth evaluation of SSWS is conducted across a number of domains to better understand the consistency in quality. The results of this evaluation are validated by repeating the procedure on a separate group of testers. Finally, an analysis of the nature of speech errors of SSWS compared to hybrid unit selection synthesis is conducted to identify the strengths and weaknesses of SSWS. Having a deeper insight into SSWS allows us to better define the focus of future work to improve this new technology.