Development of End-to-End Models: Difference between revisions

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YiningLei, Xinyi Ma, Liqing, Jingwen Shi
YiningLei, Xinyi Ma, Qing Li, Jingwen Shi


== Introduction ==
== Introduction ==
Line 62: Line 62:


== Future research ==
== Future research ==
Lorem ipsum dolor sit amet, consectetur adipiscing elit, sed do eiusmod tempor incididunt ut labore et dolore magna aliqua. Ut enim ad minim veniam, quis nostrud exercitation ullamco laboris nisi ut aliquip ex ea commodo consequat. Duis aute irure dolor in reprehenderit in voluptate velit esse cillum dolore eu fugiat nulla pariatur. Excepteur sint occaecat cupidatat non proident, sunt in culpa qui officia deserunt mollit anim id est laborum. <ref>Feng, S., Kudina, O., Halpern, B. M., & Scharenborg, O. (2021). ''Quantifying Bias in Automatic Speech Recognition'' (arXiv:2103.15122). arXiv. http://arxiv.org/abs/2103.15122</ref>
The evolving end-to-end modeling of speech recognition signals a path of exploration and innovation that is full of promise. In this section, several pivotal areas of prospective research would be illustrate that stand as linchpins for the advancement of end-to-end models in future speech recognition research.  These emergent directions encompass issues ranging from robustness in adverse acoustic environments to the ethical and privacy considerations intrinsic to widespread deployment. Furthermore, they encompass the quest for enhanced adaptability, the quest for multimodal integration, and the unceasing pursuit of ever-more accurate and contextually attuned transcriptions. Through the prism of these future research trajectories, we aim to distill the emergent imperatives that will undergird the next phase of innovation in end-to-end speech recognition models.
 
=== 1. Robustness in Noisy Acoustic Environments: ===
 
* A paramount concern in ASR research pertains to fortifying models against the deleterious effects of acoustic perturbations, particularly in real-world scenarios characterized by extraneous noise sources. Investigative endeavors may concentrate on techniques for enhancing the resilience of end-to-end ASR systems vis-à-vis a spectrum of ambient acoustic environments.
 
=== 2. Low-Resource and Under-Resourced Linguistic Contexts: ===
 
* Cognizant of the dearth of annotated data available for training ASR models in linguistically marginalized or low-resource dialects, prospective research endeavors may be directed toward methodologies that ameliorate ASR performance in resource-scarce linguistic domains. This may encompass the judicious utilization of transfer learning paradigms from well-resourced languages, in addition to unsupervised or semi-supervised learning frameworks.
 
=== 3. Adaptability and Personalization: ===
 
* Paramount to the maturation of ASR systems is the ability to tailor these models to idiosyncratic users or specialized domains, as may be pertinent in medical, legal, or other professional contexts. Areas of scholarly inquiry could span fine-tuning strategies predicated on user-specific corpora, as well as the development of techniques for adaptive model adaptation.
 
=== 4. Multimodal Convergence: ===
 
* The confluence of diverse modalities, such as audio, visual, and textual cues, embodies an incipient frontier in ASR research. Forward-looking investigations may be focused on the conceptualization and realization of end-to-end models adept at assimilating and fusing information gleaned from these heterogeneous sources to engender a more holistic comprehension of the targeted input.
 
=== 5. Continual Learning Paradigms and Incremental Training Schemes: ===
 
* Ensuring the adaptability of ASR systems to evolving contexts and the capacity for incremental knowledge acquisition is a tenet of considerable import. Research efforts in this vein may revolve around the conception of models endowed with the facility for continual learning, thereby enabling them to accrete expertise over time without necessitating wholesale retraining.
 
=== 6. Ethical and Privacy Implications: ===
 
* With the ubiquity of ASR technologies, ethical considerations loom large. Research endeavors may encompass the formulation of privacy-preserving ASR methodologies, as well as the ethical ramifications attendant to data collection, mitigation of biases, and the assurance of fairness in ASR system operation.<ref>Feng, S., Kudina, O., Halpern, B. M., & Scharenborg, O. (2021). ''Quantifying Bias in Automatic Speech Recognition'' (arXiv:2103.15122). arXiv. http://arxiv.org/abs/2103.15122</ref>
 
=== 7. Multilingual and Cross-Lingual ASR: ===
 
* The imperative to cultivate models capable of comprehending and transcribing diverse linguistic repertoires is manifestly critical in an increasingly globalized milieu. This research domain will attend to methodological frameworks that grapple with code-switching, dialectal variance, and other linguistic intricacies.
 
=== 8. Interpretability and Explicability of Models: ===
 
* Interrogating the rationales underpinning model predictions constitutes an essential endeavor, particularly in high-stakes contexts such as healthcare or legal transcription. Future research might orient itself toward the development of techniques that render end-to-end ASR models more amenable to interpretability.
 
=== 9. Zero-Shot and Few-Shot Learning Paradigms: ===
 
* Investigation into the development of models capable of generalizing to novel, unobserved tasks or languages with scant training data is a paramount research trajectory. Zero-shot and few-shot learning paradigms are likely to be pivotal in achieving this laudable objective.
 
=== 10. Deployment and Practical Applicability: ===
 
* Research efforts will be dedicated to the operationalization of end-to-end ASR systems in real-world contexts, necessitating considerations regarding computational efficiency, latency constraints, and adaptability to specific application domains.
 
These prospective avenues of inquiry underscore the unfolding trajectory of end-to-end ASR research, emblematic of a concerted effort to address salient real-world challenges and to harness the full potential of this technology.


== References ==
== References ==
To insert a reference, type <nowiki><ref> and paste the source your exported from Zotero (or whatever reference manager you're using) in the pop up box which appears. Make sure links in citations are clickable using proper formatting. Once you do this, a footnote will appear.</nowiki><ref>Glantz, Richard "SHOEBOX: a personal file handling system for textual data." In Proceedings of the November 17-19, 1970, Fall Joint Computer Conference 1970. 535-545. [https://dl.acm.org/doi/abs/10.1145/1478462.1478541]</ref> and a reference comes at the end automatically. Please use this method to cite for Wiki articles only, not for your thesis.<references /><nowiki><ref></nowiki>
To insert a reference, type <nowiki><ref> and paste the source your exported from Zotero (or whatever reference manager you're using) in the pop up box which appears. Make sure links in citations are clickable using proper formatting. Once you do this, a footnote will appear.</nowiki><ref>Glantz, Richard "SHOEBOX: a personal file handling system for textual data." In Proceedings of the November 17-19, 1970, Fall Joint Computer Conference 1970. 535-545. [https://dl.acm.org/doi/abs/10.1145/1478462.1478541]</ref> and a reference comes at the end automatically. Please use this method to cite for Wiki articles only, not for your thesis.<references /><nowiki><ref></nowiki>

Revision as of 15:32, 17 September 2023

YiningLei, Xinyi Ma, Qing Li, Jingwen Shi

Introduction

The development of End-to-End (E2E) models represent a significant shift in the field of automatic speech recognition (ASR), which seek to simplify the complex pipeline of traditional systems by directly mapping input audio sequence to sequence of words or other graphemes.[1] Framed in the Deep Learning context and taking advantage of Neural Network(NN) architectures, these models directly capture the acoustic and linguistic information present in the speech signal, casting a possibly complex processing pipeline into the coherent and flexible modeling language of neural networks.[2] The functional structure of E2E models is shown below

L = {,···,}  output sequence

Decoder

Aligner

F = {,···,}  feature sequence

Encoder

X = {,···,}  input sequence

There are several major advantages of E2E models over traditional hybrid models. First, E2E models use a single objective function which is consistent width the ASR objective to optimize the whole network, while traditional hybrid models optimize individual components separately, which cannot guarantee the global optimum. Second, E2E models perform well without deep knowledge about the problem, despite its complexity, by using a unified Neural Network architecture and an appropriate learning algorithm for Natural Language Processing (NLP), task-specific engineering and lots of prior knowledge required in traditional hybrid models can be avoided. Third, because a single network is used for ASR, E2E models are much more compact than traditional hybrid models. Therefore, E2E models has the potential to improve accuracy and efficiency in various applications, including voice assistants, transcription services, and more.

Recurrent Neural Networks (RNNs) and Convolutional Neural Networks (CNNs) have played key roles in the development of E2E models.

Historical Context

For a long time, the hidden Markov model (HMM)-Gaussian mixed model (GMM) has been the mainstream speech recognition framework. But when????, HMM-deep neural network (DNN) model and the end-to-end model using deep learning has achieved performance beyond HMM-GMM. Both using deep learning techniques, these two models have comparable performances. However, the HMM-DNN model itself is limited by various unfavorable factors such as data forced segmentation alignment, independent hypothesis, and multi-module individual training inherited from HMM, while the end-to-end model has a simplified model, joint training, direct output, no need to force data alignment and other advantages. Therefore, the end-to-end models has become a hot topic as well as important research direction in the field of speech recognition.

Traditional ASR systems involve multiple stages, including feature extraction, acoustic modeling, phonetic decoding, and language modeling. These stages often require handcrafted engineering and are computationally expensive.E2E models emerged as a response to streamline this process and leverage deep learning techniques to directly map audio to text, by replacing the original chain for a single Neural Network(NN), allowing the use of a single optimization criterion for enhancing the system.

Key Innovations

E2E modeling can directly translate the speech input into the output only using a single neural network, unlike the traditional one, which has several independent elements.

In traditional ASR, the majority of ASR systems comprise distinct acoustic, pronunciation, and language model components, each trained separately. The creation of a pronunciation lexicon and the specification of phoneme sets for a specific language necessitate expertise and are time-intensive tasks. ([2]) shows its structure.

E2E (Click here to view the E2E workflow) speech recognition significantly simplifies the complexity of traditional models. Manual labeling of information is unnecessary, as the neural network can autonomously learn language and pronunciation details.[3]

Advantage of E2E ASR:

Initially, E2E models simplify the ASR pipeline substantially by directly generating characters or even words. Conversely, the design of traditional hybrid models is intricate, demanding extensive expertise and years of ASR experience.

Furthermore, the utilization of a single network for ASR makes E2E models significantly more compact compared to traditional hybrid models. This compactness enables easy deployment of E2E models on high-accuracy devices.[4]

Last, E2E has a much simpler training approach with models, which reduces learning time, decoding time, and allows joint optimization with subsequent processing, such as understanding the natural language.[5]

Main structures for E2E speech recognition:
  • CTC

CTC as a technique to train an acoustic model without the need for precise frame-level alignments. Initially, using CTC to generate target phonemes didn't truly constitute an end-to-end method, as it still relied on language models. CTC allows for training an acoustic model without the necessity of frame-level alignments aligning acoustic data with the transcriptions.

The utilization of CTC as the loss function in training the acoustic model represents an end-to-end training approach. This method eliminates the necessity for prior data alignment, requiring only an input sequence and an output sequence for training, as illustrated in Figure 3. Consequently, manual alignment and labeling of data become unnecessary. Moreover, CTC's direct output sequence prediction doesn't require external post-processing. Within the CTC framework, 'blank' is introduced [3] (denoting no predicted value for a given frame). Each prediction's classification corresponds to a spike in the speech waveform, while non-spike regions are considered 'blank.' In a speech sequence, CTC ultimately generates a spike sequence, irrespective of each phoneme's duration.

  • Attention model

Attention-based Encoder-Decoder Models made their initial appearance within the domain of neural machine translation. The primary purpose of the Attention Mechanism is to address issues present in traditional RNN-based Seq2Seq models. A Seq2Seq model constitutes an end-to-end machine translation model, comprising an encoder and a decoder. The encoder transforms input X into a fixed-length hidden vector Z, while the decoder generates the target output Y based on Z.

Impact

Lorem ipsum dolor sit amet, consectetur adipiscing elit, sed do eiusmod tempor incididunt ut labore et dolore magna aliqua. Ut enim ad minim veniam, quis nostrud exercitation ullamco laboris nisi ut aliquip ex ea commodo consequat. Duis aute irure dolor in reprehenderit in voluptate velit esse cillum dolore eu fugiat nulla pariatur. Excepteur sint occaecat cupidatat non proident, sunt in culpa qui officia deserunt mollit anim id est laborum.

Future research

The evolving end-to-end modeling of speech recognition signals a path of exploration and innovation that is full of promise. In this section, several pivotal areas of prospective research would be illustrate that stand as linchpins for the advancement of end-to-end models in future speech recognition research. These emergent directions encompass issues ranging from robustness in adverse acoustic environments to the ethical and privacy considerations intrinsic to widespread deployment. Furthermore, they encompass the quest for enhanced adaptability, the quest for multimodal integration, and the unceasing pursuit of ever-more accurate and contextually attuned transcriptions. Through the prism of these future research trajectories, we aim to distill the emergent imperatives that will undergird the next phase of innovation in end-to-end speech recognition models.

1. Robustness in Noisy Acoustic Environments:

  • A paramount concern in ASR research pertains to fortifying models against the deleterious effects of acoustic perturbations, particularly in real-world scenarios characterized by extraneous noise sources. Investigative endeavors may concentrate on techniques for enhancing the resilience of end-to-end ASR systems vis-à-vis a spectrum of ambient acoustic environments.

2. Low-Resource and Under-Resourced Linguistic Contexts:

  • Cognizant of the dearth of annotated data available for training ASR models in linguistically marginalized or low-resource dialects, prospective research endeavors may be directed toward methodologies that ameliorate ASR performance in resource-scarce linguistic domains. This may encompass the judicious utilization of transfer learning paradigms from well-resourced languages, in addition to unsupervised or semi-supervised learning frameworks.

3. Adaptability and Personalization:

  • Paramount to the maturation of ASR systems is the ability to tailor these models to idiosyncratic users or specialized domains, as may be pertinent in medical, legal, or other professional contexts. Areas of scholarly inquiry could span fine-tuning strategies predicated on user-specific corpora, as well as the development of techniques for adaptive model adaptation.

4. Multimodal Convergence:

  • The confluence of diverse modalities, such as audio, visual, and textual cues, embodies an incipient frontier in ASR research. Forward-looking investigations may be focused on the conceptualization and realization of end-to-end models adept at assimilating and fusing information gleaned from these heterogeneous sources to engender a more holistic comprehension of the targeted input.

5. Continual Learning Paradigms and Incremental Training Schemes:

  • Ensuring the adaptability of ASR systems to evolving contexts and the capacity for incremental knowledge acquisition is a tenet of considerable import. Research efforts in this vein may revolve around the conception of models endowed with the facility for continual learning, thereby enabling them to accrete expertise over time without necessitating wholesale retraining.

6. Ethical and Privacy Implications:

  • With the ubiquity of ASR technologies, ethical considerations loom large. Research endeavors may encompass the formulation of privacy-preserving ASR methodologies, as well as the ethical ramifications attendant to data collection, mitigation of biases, and the assurance of fairness in ASR system operation.[6]

7. Multilingual and Cross-Lingual ASR:

  • The imperative to cultivate models capable of comprehending and transcribing diverse linguistic repertoires is manifestly critical in an increasingly globalized milieu. This research domain will attend to methodological frameworks that grapple with code-switching, dialectal variance, and other linguistic intricacies.

8. Interpretability and Explicability of Models:

  • Interrogating the rationales underpinning model predictions constitutes an essential endeavor, particularly in high-stakes contexts such as healthcare or legal transcription. Future research might orient itself toward the development of techniques that render end-to-end ASR models more amenable to interpretability.

9. Zero-Shot and Few-Shot Learning Paradigms:

  • Investigation into the development of models capable of generalizing to novel, unobserved tasks or languages with scant training data is a paramount research trajectory. Zero-shot and few-shot learning paradigms are likely to be pivotal in achieving this laudable objective.

10. Deployment and Practical Applicability:

  • Research efforts will be dedicated to the operationalization of end-to-end ASR systems in real-world contexts, necessitating considerations regarding computational efficiency, latency constraints, and adaptability to specific application domains.

These prospective avenues of inquiry underscore the unfolding trajectory of end-to-end ASR research, emblematic of a concerted effort to address salient real-world challenges and to harness the full potential of this technology.

References

To insert a reference, type <ref> and paste the source your exported from Zotero (or whatever reference manager you're using) in the pop up box which appears. Make sure links in citations are clickable using proper formatting. Once you do this, a footnote will appear.[7] and a reference comes at the end automatically. Please use this method to cite for Wiki articles only, not for your thesis.

  1. Wang, Dong, Xiaodong Wang, and Shaohe Lv. 2019. “An Overview of End-to-End Automatic Speech Recognition.” Symmetry 11(8):1018.
  2. Glasmachers, Tobias. “Limits of end-to-end learning.”arXiv preprint arXiv:1704.08305 (2017).
  3. Wang, Dong, Xiaodong Wang, and Shaohe Lv. 2019. “An Overview of End-to-End Automatic Speech Recognition.” Symmetry 11(8):1018.
  4. Li J .Recent Advances in End-to-End Automatic Speech Recognition[J]. 2021.DOI:10.48550/arXiv.2111.01690.
  5. Orken M , Dina O , Keylan A ,et al.A study of transformer-based end-to-end speech recognition system for Kazakh language[J].Scientific Reports[2023-09-17].DOI:10.1038/s41598-022-12260-y.
  6. Feng, S., Kudina, O., Halpern, B. M., & Scharenborg, O. (2021). Quantifying Bias in Automatic Speech Recognition (arXiv:2103.15122). arXiv. http://arxiv.org/abs/2103.15122
  7. Glantz, Richard "SHOEBOX: a personal file handling system for textual data." In Proceedings of the November 17-19, 1970, Fall Joint Computer Conference 1970. 535-545. [1]

<ref>