keyword
phoneme classification
Phoneme classification is a supervised machine learning task in speech processing that involves assigning the correct phoneme label to a pre-segmented acoustic frame or segment of spoken audio. Because phonemes represent the smallest contrastive units of sound that distinguish words in a language, this task requires a model to map acoustic features, such as spectral and temporal patterns, to discrete phonetic categories. Unlike continuous phoneme recognition, which must jointly infer time boundaries and decode continuous speech sequences, phoneme classification assumes pre-determined boundary locations and evaluates how effectively an acoustic model can discriminate phonetic content in isolation. It is widely used as a standard benchmark to evaluate acoustic feature extractors, deep neural network architectures, and self-supervised speech representations on their capacity to capture fine-grained phonetic information while remaining robust to variations in speaker identity, pitch, and acoustic environment.
3 items

Unified Speech-Text Pre-training for Speech Translation and Recognition
Yun Tang, Hongyu Gong, Ning Dong, Changhan Wang, Wei-Ning Hsu, Jiatao Gu, Alexei Baevski, Xian Li, Abdelrahman Mohamed, Michael Auli, Juan Miguel Pino
Why you should read this
Proposes a joint speech-text pre-training framework that integrates linguistic knowledge from text into speech models across four multi-task objectives, providing tailored encoder sharing strategies to resolve subtask interference and substantially boosting performance on speech translation and recognition.
We describe a method to jointly pre-train speech and text in an encoder-decoder modeling framework for speech translation and recognition. The proposed method incorporates four self-supervised and supervised subtasks for cross modality learning. A self-supervised speech subtask leverages unlabelled speech data, and a (self-)supervised text to text subtask makes use of abundant text training data. Two auxiliary supervised speech tasks are included to unify speech and text modeling space. Our contribution lies in integrating linguistic information from the text corpus into the speech pre-training. Detailed analysis reveals learning interference among subtasks. Two pre-training configurations for speech translation and recognition, respectively, are presented to alleviate subtask interference. Our experiments show the proposed method can effectively fuse speech and text information into one model. It achieves between 1.7 and 2.3 BLEU improvement above the state of the art on the MUST-C speech translation dataset and comparable WERs to wav2vec 2.0 on the LIBRISPEECH speech recognition task.
Added
2026-09-26

Fundamentals of Recurrent Neural Network (RNN) and Long Short-Term Memory (LSTM) Network
Alex Sherstinsky
Why you should read this
Establishes a rigorous mathematical foundation for recurrent neural networks and LSTMs by deriving their formulations from differential equations, formally proving the unrolling method, and providing complete training equations alongside a generalized model variant.
Because of their effectiveness in broad practical applications, LSTM networks have received a wealth of coverage in scientific journals, technical blogs, and implementation guides. However, in most articles, the inference formulas for the LSTM network and its parent, RNN, are stated axiomatically, while the training formulas are omitted altogether. In addition, the technique of "unrolling" an RNN is routinely presented without justification throughout the literature. The goal of this paper is to explain the essential RNN and LSTM fundamentals in a single document. Drawing from concepts in signal processing, we formally derive the canonical RNN formulation from differential equations. We then propose and prove a precise statement, which yields the RNN unrolling technique. We also review the difficulties with training the standard RNN and address them by transforming the RNN into the "Vanilla LSTM" network through a series of logical arguments. We provide all equations pertaining to the LSTM system together with detailed descriptions of its constituent entities. Albeit unconventional, our choice of notation and the method for presenting the LSTM system emphasizes ease of understanding. As part of the analysis, we identify new opportunities to enrich the LSTM system and incorporate these extensions into the Vanilla LSTM network, producing the most general LSTM variant to date. The target reader has already been exposed to RNNs and LSTM networks through numerous available resources and is open to an alternative pedagogical approach. A Machine Learning practitioner seeking guidance for implementing our new augmented LSTM model in software for experimentation and research will find the insights and derivations in this tutorial valuable as well.
Added
2026-09-14

WavLM: Large-Scale Self-Supervised Pre-Training for Full Stack Speech Processing
Sanyuan Chen, Chengyi Wang, Zhengyang Chen, Yu Wu, Shujie Liu, Zhuo Chen, Jinyu Li, Naoyuki Kanda, Takuya Yoshioka, Xiong Xiao, Jian Wu, Long Zhou, Shuo Ren, Y. Qian, Yao Qian, Micheal Zeng, Furu Wei
Why you should read this
Introduces WavLM, a self-supervised speech foundation model that combines masked speech modeling with denoising on 94,000 hours of audio to achieve state-of-the-art performance across diverse full-stack speech tasks on the SUPERB benchmark.
Self-supervised learning (SSL) achieves great success in speech recognition, while limited exploration has been attempted for other speech processing tasks. As speech signal contains multi-faceted information including speaker identity, paralinguistics, spoken content, etc., learning universal representations for all speech tasks is challenging. To tackle the problem, we propose a new pre-trained model, WavLM, to solve full-stack downstream speech tasks. WavLM jointly learns masked speech prediction and denoising in pre-training. By this means, WavLM does not only keep the speech content modeling capability by the masked speech prediction, but also improves the potential to non-ASR tasks by the speech denoising. In addition, WavLM employs gated relative position bias for the Transformer structure to better capture the sequence ordering of input speech. We also scale up the training dataset from 60k hours to 94k hours. WavLM Large achieves state-of-the-art performance on the SUPERB benchmark, and brings significant improvements for various speech processing tasks on their representative benchmarks. The code and pre-trained models are available at this https URL.
Added
2026-09-13
