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In this study, we investigate the integration of a large language model (LLM) with an automatic speech recognition (ASR) system, specifically focusing on enhancing rare word recognition performance. Using a 190,000-hour dataset primarily…

计算与语言 · 计算机科学 2025-02-25 Haoxuan Wang

Error correction (EC) based on large language models is an emerging technology to enhance the performance of automatic speech recognition (ASR) systems. Generally, training data for EC are collected by automatically pairing a large set of…

计算与语言 · 计算机科学 2024-10-17 Takuma Udagawa , Masayuki Suzuki , Masayasu Muraoka , Gakuto Kurata

This paper describes AaltoASR's speech recognition system for the INTERSPEECH 2020 shared task on Automatic Speech Recognition (ASR) for non-native children's speech. The task is to recognize non-native speech from children of various age…

音频与语音处理 · 电气工程与系统科学 2020-09-01 Hemant Kathania , Mittul Singh , Tamás Grósz , Mikko Kurimo

This paper presents a new challenge that calls for zero-shot text-to-speech (TTS) systems to augment speech data for the downstream task, personalized speech enhancement (PSE), as part of the Generative Data Augmentation workshop at ICASSP…

音频与语音处理 · 电气工程与系统科学 2025-01-24 Jae-Sung Bae , Anastasia Kuznetsova , Dinesh Manocha , John Hershey , Trausti Kristjansson , Minje Kim

Research on automatic speech recognition (ASR) systems for electrolaryngeal speakers has been relatively unexplored due to small datasets. When training data is lacking in ASR, a large-scale pretraining and fine tuning framework is often…

声音 · 计算机科学 2023-05-31 Lester Phillip Violeta , Ding Ma , Wen-Chin Huang , Tomoki Toda

In the realm of automatic speech recognition (ASR), robustness in noisy environments remains a significant challenge. Recent ASR models, such as Whisper, have shown promise, but their efficacy in noisy conditions can be further enhanced.…

声音 · 计算机科学 2024-06-28 Yehoshua Dissen , Shiry Yonash , Israel Cohen , Joseph Keshet

Developing Automatic Speech Recognition (ASR) for low-resource languages is a challenge due to the small amount of transcribed audio data. For many such languages, audio and text are available separately, but not audio with transcriptions.…

计算与语言 · 计算机科学 2022-07-21 Nathaniel Robinson , Perez Ogayo , Swetha Gangu , David R. Mortensen , Shinji Watanabe

Automatic Speech Recognition (ASR) plays a crucial role in human-machine interaction and serves as an interface for a wide range of applications. Traditionally, ASR performance has been evaluated using Word Error Rate (WER), a metric that…

音频与语音处理 · 电气工程与系统科学 2025-07-23 Sujith Pulikodan , Sahapthan K , Prasanta Kumar Ghosh , Visruth Sanka , Nihar Desai

Automatic Speech Recognition (ASR) models have achieved remarkable accuracy in general settings, yet their performance often degrades in domain-specific applications due to data mismatch and linguistic variability. This challenge is…

Data augmentation is a technique to generate new training data based on existing data. We evaluate the simple and cost-effective method of concatenating the original data examples to build new training instances. Continued training with…

计算与语言 · 计算机科学 2023-06-12 Tsz Kin Lam , Shigehiko Schamoni , Stefan Riezler

Speech synthesis technology has witnessed significant advancements in recent years, enabling the creation of natural and expressive synthetic speech. One area of particular interest is the generation of synthetic child speech, which…

声音 · 计算机科学 2023-11-09 Rishabh Jain , Peter Corcoran

Large language model (LLM)-based automatic speech recognition (ASR) has recently achieved strong performance across diverse tasks, yet contextual biasing for named entities and hotwords under large vocabularies remains challenging. In this…

音频与语音处理 · 电气工程与系统科学 2025-12-29 YuXiang Kong , JunFeng Hou , Jian Tang , Bingqing Zhu , Jicheng Zhang , Shaofei Xue

Automatic speech recognition (ASR) systems have achieved strong performance on general transcription tasks. However, they continue to struggle with recognizing rare named entities and adapting to domain mismatches. In contrast, large…

计算与语言 · 计算机科学 2025-08-21 Shaoshi Ling , Guoli Ye

We recently developed SLM, a joint speech and language model, which fuses a pretrained foundational speech model and a large language model (LLM), while preserving the in-context learning capability intrinsic to the pretrained LLM. In this…

计算与语言 · 计算机科学 2024-02-08 Mingqiu Wang , Izhak Shafran , Hagen Soltau , Wei Han , Yuan Cao , Dian Yu , Laurent El Shafey

This paper proposes an approach to build a high-quality text-to-speech (TTS) system for technical domains using data augmentation. An end-to-end (E2E) system is trained on hidden Markov model (HMM) based synthesized speech and further…

音频与语音处理 · 电气工程与系统科学 2022-12-23 Ishika Gupta , Anusha Prakash , Jom Kuriakose , Hema A. Murthy

We introduce a new cross-modal fusion technique designed for generative error correction in automatic speech recognition (ASR). Our methodology leverages both acoustic information and external linguistic representations to generate accurate…

Data augmentation is commonly used to help build a robust speaker verification system, especially in limited-resource case. However, conventional data augmentation methods usually focus on the diversity of acoustic environment, leaving the…

声音 · 计算机科学 2021-02-22 Houjun Huang , Xu Xiang , Fei Zhao , Shuai Wang , Yanmin Qian

Dysarthric speech reconstruction (DSR) typically employs a cascaded system that combines automatic speech recognition (ASR) and sentence-level text-to-speech (TTS) to convert dysarthric speech into normally-prosodied speech. However,…

声音 · 计算机科学 2026-03-03 Minghui Wu , Haitao Tang , Jiahuan Fan , Ruizhi Liao , Yanyong Zhang

As more and more information-rich data like video become available, utilizing multi-modal auxiliary information to enhance audio tasks has sparked widespread research interest. The recent surge in research on LLM-based audio models provides…

音频与语音处理 · 电气工程与系统科学 2024-11-12 Guanrou Yang , Ziyang Ma , Fan Yu , Zhifu Gao , Shiliang Zhang , Xie Chen

This study investigates the impact of integrating a dataset of disordered speech recordings ($\sim$1,000 hours) into the fine-tuning of a near state-of-the-art ASR baseline system. Contrary to what one might expect, despite the data being…

音频与语音处理 · 电气工程与系统科学 2025-12-22 Jimmy Tobin , Katrin Tomanek , Subhashini Venugopalan