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OpenAI's Whisper Automated Speech Recognition model excels in generalizing across diverse datasets and domains. However, this broad adaptability can lead to diminished performance in tasks requiring recognition of specific vocabularies.…

人工智能 · 计算机科学 2025-08-12 Vishakha Lall , Yisi Liu

Whisper generation is constrained by the difficulty of data collection. Because whispered speech has low acoustic amplitude, high-fidelity recording is challenging. In this paper, we introduce WhispSynth, a large-scale multilingual corpus…

声音 · 计算机科学 2026-03-17 Tianyi Tan , Jiaxin Ye , Yuanming Zhang , Xiaohuai Le , Xianjun Xia , Chuanzeng Huang , Jing Lu

This paper tackles several challenges that arise when integrating Automatic Speech Recognition (ASR) and Machine Translation (MT) for real-time, on-device streaming speech translation. Although state-of-the-art ASR systems based on…

计算与语言 · 计算机科学 2025-08-20 Zeeshan Ahmed , Frank Seide , Niko Moritz , Ju Lin , Ruiming Xie , Simone Merello , Zhe Liu , Christian Fuegen

An inferior performance of the streaming automatic speech recognition models versus non-streaming model is frequently seen due to the absence of future context. In order to improve the performance of the streaming model and reduce the…

声音 · 计算机科学 2022-03-30 Jingyu Sun , Guiping Zhong , Dinghao Zhou , Baoxiang Li

Token-based language modeling is a prominent approach for speech generation, where tokens are obtained by quantizing features from self-supervised learning (SSL) models and extracting codes from neural speech codecs, generally referred to…

音频与语音处理 · 电气工程与系统科学 2025-06-30 Yang Yang , Yunpeng Li , George Sung , Shao-Fu Shih , Craig Dooley , Alessio Centazzo , Ramanan Rajeswaran

Recognizing whispered speech and converting it to normal speech creates many possibilities for speech interaction. Because the sound pressure of whispered speech is significantly lower than that of normal speech, it can be used as a…

声音 · 计算机科学 2023-03-06 Jun Rekimoto

Deploying modern Speech Language Models (SpeechLMs) in streaming settings requires systems that provide low latency, high throughput, and strong guarantees of streamability. Existing systems fall short of supporting diverse models flexibly…

In this work we propose an inference technique, asynchronous revision, to unify streaming and non-streaming speech recognition models. Specifically, we achieve dynamic latency with only one model by using arbitrary right context during…

音频与语音处理 · 电气工程与系统科学 2020-11-04 Mingkun Huang , Meng Cai , Jun Zhang , Yang Zhang , Yongbin You , Yi He , Zejun Ma

Encoder-decoder models have achieved remarkable success in speech and text tasks, yet efficiently adapting these models to diverse uni/multi-modal scenarios remains an open challenge. In this paper, we propose Whisper-UT, a unified and…

User studies have shown that reducing the latency of our simultaneous lecture translation system should be the most important goal. We therefore have worked on several techniques for reducing the latency for both components, the automatic…

音频与语音处理 · 电气工程与系统科学 2020-03-24 Thai Son Nguyen , Jan Niehues , Eunah Cho , Thanh-Le Ha , Kevin Kilgour , Markus Muller , Matthias Sperber , Sebastian Stueker , Alex Waibel

Fast Automatic Speech Recognition (ASR) is critical for latency-sensitive applications such as real-time captioning and meeting transcription. However, truly parallel ASR decoding remains challenging due to the sequential nature of…

We propose a novel approach to enable the use of large, single-speaker ASR models, such as Whisper, for target speaker ASR. The key claim of this method is that it is much easier to model relative differences among speakers by learning to…

音频与语音处理 · 电气工程与系统科学 2025-01-17 Alexander Polok , Dominik Klement , Matthew Wiesner , Sanjeev Khudanpur , Jan Černocký , Lukáš Burget

Recent advances in zero-shot text-to-speech (TTS) synthesis have achieved high-quality speech generation for unseen speakers, but most systems remain unsuitable for real-time applications because of their offline design. Current streaming…

声音 · 计算机科学 2025-06-17 Hui Wang , Yifan Yang , Shujie Liu , Jinyu Li , Lingwei Meng , Yanqing Liu , Jiaming Zhou , Haoqin Sun , Yan Lu , Yong Qin

The recently-developed WaveNet architecture is the current state of the art in realistic speech synthesis, consistently rated as more natural sounding for many different languages than any previous system. However, because WaveNet relies on…

This paper introduces a fast-slow encoder based transducer with streaming deliberation for end-to-end automatic speech recognition. We aim to improve the recognition accuracy of the fast-slow encoder based transducer while keeping its…

音频与语音处理 · 电气工程与系统科学 2022-12-16 Ke Li , Jay Mahadeokar , Jinxi Guo , Yangyang Shi , Gil Keren , Ozlem Kalinli , Michael L. Seltzer , Duc Le

With the increased audiovisualisation of communication, the need for live subtitles in multilingual events is more relevant than ever. In an attempt to automatise the process, we aim at exploring the feasibility of simultaneous speech…

计算与语言 · 计算机科学 2023-10-19 Alina Karakanta , Sara Papi , Matteo Negri , Marco Turchi

Streaming multi-talker speech translation is a task that involves not only generating accurate and fluent translations with low latency but also recognizing when a speaker change occurs and what the speaker's gender is. Speaker change…

As a popular multilingual and multitask pre-trained speech model, Whisper has the problem of curse of multilinguality. To enhance multilingual capabilities in small Whisper models, we propose DQ-Whisper, a novel joint distillation and…

声音 · 计算机科学 2024-10-01 Hang Shao , Bei Liu , Wei Wang , Xun Gong , Yanmin Qian

Large encoder-decoder models like Whisper achieve strong offline transcription but remain impractical for streaming applications due to high latency. However, due to the accessibility of pre-trained checkpoints, the open Thai ASR landscape…

Speaker verification is a task of confirming an individual's identity through the analysis of their voice. Whispered speech differs from phonated speech in acoustic characteristics, which degrades the performance of speaker verification…

声音 · 计算机科学 2026-05-08 Magdalena Gołębiowska , Piotr Syga