The DOI matches its Crossref record; 33 of the 33 references it cites were found in a registry or an index.
Conformer: Convolution-augmented Transformer for Speech Recognition
Open accessAbstract
Recently Transformer and Convolution neural network (CNN) based models have shown promising results in Automatic Speech Recognition (ASR), outperforming Recurrent neural networks (RNNs). Transformer models are good at capturing content-based global interactions, while CNNs exploit local features effectively. In this work, we achieve the best of both worlds by studying how to combine convolution neural networks and transformers to model both local and global dependencies of an audio sequence in a parameter-efficient way. To this regard, we propose the convolution-augmented transformer for speech recognition, named Conformer. Conformer significantly outperforms the previous Transformer and CNN based models achieving state-of-the-art accuracies. On the widely used LibriSpeech benchmark, our model achieves WER of 2.1%/4.3% without using a language model and 1.9%/3.9% with an external language model on test/testother. We also observe competitive performance of 2.7%/6.3% with a small model of only 10M parameters.
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“In particular, transformer1 and recurrent neural-network transducer (RNNT)12,13,16 models containing up to one billion parameters2 have produced a marked decrease in word error rate (WER) (and therefore much better accuracy) for the automated transcription of spoken English-language sentences, as shown in Fig. 1a for two widely used datasets, Librispeech17 and SwitchBoard18.”
From An analog-AI chip for energy-efficient speech recognition and transcription (2023) · cites this paper as [16]
In the citing paper
“A range of acoustic models that give state-of-the-art performance on these tasks are chosen as the baseline speech recognition systems, including hybrid DNN [41], hybrid lattice-free maximum mutual information (LF-MMI) trained time delay neural network (TDNN) [3] and end-to-end (E2E) Conformer [6] models.”
From Speaker adaptation using spectro-temporal deep features for dysarthric and elderly speech recognition (2022) · cites this paper as [6]
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Metadata
- Authors
- Gulati, A., Qin, J., Chiu, C.-C., Parmar, N., Zhang, Y., Yu, J., Han, W., Wang, S., Zhang, Z., Wu, Y., & Pang, R.
- Affiliations
- Google (United States) (US)
- ORCID
- Venue
- Interspeech 2020
- Publisher
- ISCA
- Published
- 2020-10-25
- Pages
- 5036-5040
- ISSN
- 2308-457X (ISSN-L)
- OpenAlex
- W3097777922
- Cited by
- 3,150
- References
- 33
- Citations by year
- 2026: 373 · 2025: 666 · 2024: 644 · 2023: 708 · 2022: 424 · 2021: 300 · 2020: 35
- FWCI
- 140.10
- Type
- conference paper
- Language
- en
- Open access
- green
- arxiv.org
- Licence
- public-domain
- Locations
- Interspeech (USB) (publishedVersion)
- arXiv (Cornell University) (submittedVersion, public-domain)
- arXiv (Cornell University)
- Topics
- Physical Sciences › Computer Science › Artificial Intelligence › Speech Recognition and Synthesis
- Physical Sciences › Computer Science › Signal Processing › Speech and Audio Processing
- Physical Sciences › Computer Science › Signal Processing › Music and Audio Processing
- Keywords
- conformational isomers, speech recognition, CNN-Transformer hybrid, parameter-efficient models, LibriSpeech, word error rate
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