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相关论文: Turning Whisper into Real-Time Transcription Syste…

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Real-time Automatic Speech Recognition (ASR) is a fundamental building block for many commercial applications of ML, including live captioning, dictation, meeting transcriptions, and medical scribes. Accuracy and latency are the most…

声音 · 计算机科学 2025-07-16 Atila Orhon , Arda Okan , Berkin Durmus , Zach Nagengast , Eduardo Pacheco

We propose DarkStream, a streaming speech synthesis model for real-time speaker anonymization. To improve content encoding under strict latency constraints, DarkStream combines a causal waveform encoder, a short lookahead buffer, and…

音频与语音处理 · 电气工程与系统科学 2025-09-08 Waris Quamer , Ricardo Gutierrez-Osuna

Streaming speech-to-text translation (StreamST) is the task of automatically translating speech while incrementally receiving an audio stream. Unlike simultaneous ST (SimulST), which deals with pre-segmented speech, StreamST faces the…

声音 · 计算机科学 2024-06-11 Sara Papi , Marco Gaido , Matteo Negri , Luisa Bentivogli

Whisper is a state-of-the-art automatic speech recognition (ASR) model (Radford et al., 2022). Although Swiss German dialects are allegedly not part of Whisper's training data, preliminary experiments showed that Whisper can transcribe…

计算与语言 · 计算机科学 2024-05-10 Eyal Liron Dolev , Clemens Fidel Lutz , Noëmi Aepli

Diffusion models are a class of generative models that have been recently used for speech enhancement with remarkable success but are computationally expensive at inference time. Therefore, these models are impractical for processing…

音频与语音处理 · 电气工程与系统科学 2025-09-15 Bunlong Lay , Rostislav Makarov , Timo Gerkmann

Lipreading has a lot of potential applications such as in the domain of surveillance and video conferencing. Despite this, most of the work in building lipreading systems has been limited to classifying silent videos into classes…

音频与语音处理 · 电气工程与系统科学 2019-07-03 Yaman Kumar , Rohit Jain , Khwaja Mohd. Salik , Rajiv Ratn Shah , Yifang yin , Roger Zimmermann

Macroscopic intelligibility models predict the expected human word-error-rate for a given speech-in-noise stimulus. In contrast, microscopic intelligibility models aim to make fine-grained predictions about listeners' perception, e.g.…

音频与语音处理 · 电气工程与系统科学 2024-09-12 Paul Best , Santiago Cuervo , Ricard Marxer

Diffusion-based generative models have greatly impacted the speech processing field in recent years, exhibiting high speech naturalness and spawning a new research direction. Their application in real-time communication is, however, still…

信号处理 · 电气工程与系统科学 2026-04-22 Simon Welker , Bunlong Lay , Maris Hillemann , Tal Peer , Timo Gerkmann

The growing need for instant spoken language transcription and translation is driven by increased global communication and cross-lingual interactions. This has made offering translations in multiple languages essential for user…

计算与语言 · 计算机科学 2023-10-24 Sara Papi , Peidong Wang , Junkun Chen , Jian Xue , Naoyuki Kanda , Jinyu Li , Yashesh Gaur

How to make human-interpreter-like read/write decisions for simultaneous speech translation (SimulST) systems? Current state-of-the-art systems formulate SimulST as a multi-turn dialogue task, requiring specialized interleaved training data…

计算与语言 · 计算机科学 2026-02-02 Haotian Tan , Hiroki Ouchi , Sakriani Sakti

Simultaneous machine translation (SiMT) has traditionally relied on offline machine translation models coupled with human-engineered heuristics or learned policies. We propose Hikari, a policy-free, fully end-to-end model that performs…

计算与语言 · 计算机科学 2026-03-13 Roman Koshkin , Jeon Haesung , Lianbo Liu , Hao Shi , Mengjie Zhao , Yusuke Fujita , Yui Sudo

Real-time object detection is critical for the decision-making process for many real-world applications, such as collision avoidance and path planning in autonomous driving. This work presents an innovative real-time streaming perception…

计算机视觉与模式识别 · 计算机科学 2024-09-11 Xiang Zhang , Yufei Cui , Chenchen Fu , Weiwei Wu , Zihao Wang , Yuyang Sun , Xue Liu

Audio-Visual Speech Recognition (AVSR) uses lip-based video to improve performance in noise. Since videos are harder to obtain than audio, the video training data of AVSR models is usually limited to a few thousand hours. In contrast,…

音频与语音处理 · 电气工程与系统科学 2024-11-21 Andrew Rouditchenko , Yuan Gong , Samuel Thomas , Leonid Karlinsky , Hilde Kuehne , Rogerio Feris , James Glass

Transformer-based neural speech processing has achieved state-of-the-art performance. Since speech audio signals are known to be highly compressible, here we seek to accelerate neural speech transcription by time-domain signal…

机器学习 · 计算机科学 2025-06-23 Zifei Xu , Sayeh Sharify , Hesham Mostafa , Tristan Webb , Wanzin Yazar , Xin Wang

A crucial part of an accurate and reliable spoken language assessment system is the underlying ASR model. Recently, large-scale pre-trained ASR foundation models such as Whisper have been made available. As the output of these models is…

计算与语言 · 计算机科学 2023-10-11 Rao Ma , Mengjie Qian , Mark J. F. Gales , Kate M. Knill

ASR systems often struggle with maintaining syntactic and semantic accuracy in long audio transcripts, impacting tasks like Named Entity Recognition (NER), capitalization, and punctuation. We propose a novel approach that enhances ASR by…

计算与语言 · 计算机科学 2025-08-20 Duygu Altinok

This research introduces an enhanced version of the multi-objective speech assessment model--MOSA-Net+, by leveraging the acoustic features from Whisper, a large-scaled weakly supervised model. We first investigate the effectiveness of…

音频与语音处理 · 电气工程与系统科学 2024-04-30 Ryandhimas E. Zezario , Yu-Wen Chen , Szu-Wei Fu , Yu Tsao , Hsin-Min Wang , Chiou-Shann Fuh

Automatic dubbing, which generates a corresponding version of the input speech in another language, could be widely utilized in many real-world scenarios such as video and game localization. In addition to synthesizing the translated…

声音 · 计算机科学 2024-07-08 Jingbei Li , Sipan Li , Ping Chen , Luwen Zhang , Yi Meng , Zhiyong Wu , Helen Meng , Qiao Tian , Yuping Wang , Yuxuan Wang

Streaming end-to-end automatic speech recognition (ASR) models are widely used on smart speakers and on-device applications. Since these models are expected to transcribe speech with minimal latency, they are constrained to be causal with…

We introduce DiffuseST, a low-latency, direct speech-to-speech translation system capable of preserving the input speaker's voice zero-shot while translating from multiple source languages into English. We experiment with the synthesizer…