# On Architectures and Training for Raw Waveform Feature Extraction in ASR

arXiv: https://arxiv.org/abs/2104.04298 · HTML: https://mlsanity.com/paper/2104.04298/

## Key facts

- Cited by 8 papers in the map (data through 14 Oct 2026).
- Builds on 16 of its 35 references that are in the map.
- First posted on arXiv 9 Apr 2021.

## Builds on (most cited first, 16 of 16)

- [wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Representations](https://mlsanity.com/paper/2006.11477/) (2020)
- [Conformer: Convolution-augmented Transformer for Speech Recognition](https://mlsanity.com/paper/2005.08100/) (2020)
- [Listen, Attend and Spell](https://mlsanity.com/paper/1508.01211/) (2015)
- [Unsupervised Cross-lingual Representation Learning for Speech Recognition](https://mlsanity.com/paper/2006.13979/) (2020)
- [vq-wav2vec: Self-Supervised Learning of Discrete Speech Representations](https://mlsanity.com/paper/1910.05453/) (2019)
- [Speaker Recognition from Raw Waveform with SincNet](https://mlsanity.com/paper/1808.00158/) (2018)
- [Hybrid Autoregressive Transducer (hat)](https://mlsanity.com/paper/2003.07705/) (2020)
- [Language Modeling with Deep Transformers](https://mlsanity.com/paper/1905.04226/) (2019)
- [RWTH ASR Systems for LibriSpeech: Hybrid vs Attention -- w/o Data Augmentation](https://mlsanity.com/paper/1905.03072/) (2019)
- [The Microsoft 2017 Conversational Speech Recognition System](https://mlsanity.com/paper/1708.06073/) (2017)
- [Learning Robust and Multilingual Speech Representations](https://mlsanity.com/paper/2001.11128/) (2020)
- [Phoneme Based Neural Transducer for Large Vocabulary Speech Recognition](https://mlsanity.com/paper/2010.16368/) (2020)
- [The RWTH ASR System for TED-LIUM Release 2: Improving Hybrid HMM with SpecAugment](https://mlsanity.com/paper/2004.00960/) (2020)
- [Cumulative Adaptation for BLSTM Acoustic Models](https://mlsanity.com/paper/1906.06207/) (2019)
- [LSTM Language Models for LVCSR in First-Pass Decoding and Lattice-Rescoring](https://mlsanity.com/paper/1907.01030/) (2019)
- [Faster, Simpler and More Accurate Hybrid ASR Systems Using Wordpieces](https://mlsanity.com/paper/2005.09150/) (2020)

## Built on by (most cited first, 8 of 8)

- [On Language Model Integration for RNN Transducer based Speech Recognition](https://mlsanity.com/paper/2110.06841/) (2021)
- [FeaRLESS: Feature Refinement Loss for Ensembling Self-Supervised Learning Features in Robust End-to-end Speech Recognition](https://mlsanity.com/paper/2206.15056/) (2022)
- [Efficient Utilization of Large Pre-Trained Models for Low Resource ASR](https://mlsanity.com/paper/2210.15445/) (2022)
- [Comparative Analysis of the wav2vec 2.0 Feature Extractor](https://mlsanity.com/paper/2308.04286/) (2023)
- [Unsupervised Pre-Training for Vietnamese Automatic Speech Recognition in the HYKIST Project](https://mlsanity.com/paper/2309.15869/) (2023)
- [Hold Me Tight: Stable Encoder-Decoder Design for Speech Enhancement](https://mlsanity.com/paper/2408.17358/) (2024)
- [Regularizing Learnable Feature Extraction for Automatic Speech Recognition](https://mlsanity.com/paper/2506.09804/) (2025)
- [Unified Learnable 2D Convolutional Feature Extraction for ASR](https://mlsanity.com/paper/2509.10031/) (2025)

## Abstract (arXiv)

With the success of neural network based modeling in automatic speech recognition (ASR), many studies investigated acoustic modeling and learning of feature extractors directly based on the raw waveform. Recently, one line of research has focused on unsupervised pre-training of feature extractors on audio-only data to improve downstream ASR performance. In this work, we investigate the usefulness of one of these front-end frameworks, namely wav2vec, in a setting without additional untranscribed data for hybrid ASR systems. We compare this framework both to the manually defined standard Gammatone feature set, as well as to features extracted as part of the acoustic model of an ASR system trained supervised. We study the benefits of using the pre-trained feature extractor and explore how to additionally exploit an existing acoustic model trained with different features. Finally, we systematically examine combinations of the described features in order to further advance the performance.
