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This paper investigates the use of target-speaker automatic speech recognition (TS-ASR) for simultaneous speech recognition and speaker diarization of single-channel dialogue recordings. TS-ASR is a technique to automatically extract and…

计算与语言 · 计算机科学 2019-09-19 Naoyuki Kanda , Shota Horiguchi , Yusuke Fujita , Yawen Xue , Kenji Nagamatsu , Shinji Watanabe

Speaker diarization, the process of segmenting an audio stream or transcribed speech content into homogenous partitions based on speaker identity, plays a crucial role in the interpretation and analysis of human speech. Most existing…

机器学习 · 计算机科学 2024-08-23 Luyao Cheng , Hui Wang , Siqi Zheng , Yafeng Chen , Rongjie Huang , Qinglin Zhang , Qian Chen , Xihao Li

An important problem in ad-hoc microphone speech separation is how to guarantee the robustness of a system with respect to the locations and numbers of microphones. The former requires the system to be invariant to different indexing of the…

音频与语音处理 · 电气工程与系统科学 2020-03-30 Yi Luo , Zhuo Chen , Nima Mesgarani , Takuya Yoshioka

High quality speech capture has been widely studied for both voice communication and human computer interface reasons. To improve the capture performance, we can often find multi-microphone speech enhancement techniques deployed on various…

音频与语音处理 · 电气工程与系统科学 2023-03-15 Yang Yang , Shao-Fu Shih , Hakan Erdogan , Jamie Menjay Lin , Chehung Lee , Yunpeng Li , George Sung , Matthias Grundmann

This paper proposes a unified deep speaker embedding framework for modeling speech data with different sampling rates. Considering the narrowband spectrogram as a sub-image of the wideband spectrogram, we tackle the joint modeling problem…

音频与语音处理 · 电气工程与系统科学 2020-12-02 Weicheng Cai , Ming Li

Multi-talker automatic speech recognition (ASR) has been studied to generate transcriptions of natural conversation including overlapping speech of multiple speakers. Due to the difficulty in acquiring real conversation data with…

音频与语音处理 · 电气工程与系统科学 2022-11-21 Muqiao Yang , Naoyuki Kanda , Xiaofei Wang , Jian Wu , Sunit Sivasankaran , Zhuo Chen , Jinyu Li , Takuya Yoshioka

Current state-of-the-art speech recognition models are trained to map acoustic signals into sub-lexical units. While these models demonstrate superior performance, they remain vulnerable to out-of-distribution conditions such as background…

声音 · 计算机科学 2024-10-10 Sagarika Alavilli , Annesya Banerjee , Gasser Elbanna , Annika Magaro

Purpose: Surgical scene understanding is key to advancing computer-aided and intelligent surgical systems. Current approaches predominantly rely on visual data or end-to-end learning, which limits fine-grained contextual modeling. This work…

This study describes a binaural machine hearing system that is capable of performing auditory stream segregation in scenarios where multiple sound sources are present. The process of stream segregation refers to the capability of human…

声音 · 计算机科学 2016-06-27 Christopher Schymura , Thomas Walther , Dorothea Kolossa

Joint audio-visual speaker tracking requires that the locations of microphones and cameras are known and that they are given in a common coordinate system. Sensor self-localization algorithms, however, are usually separately developed for…

声音 · 计算机科学 2015-04-14 Florian Jacob , Reinhold Haeb-Umbach

In this work we explored building automatic speech recognition models for transcribing doctor patient conversation. We collected a large scale dataset of clinical conversations ($14,000$ hr), designed the task to represent the real word…

Identification and localization of sounds are both integral parts of computational auditory scene analysis. Although each can be solved separately, the goal of forming coherent auditory objects and achieving a comprehensive spatial scene…

声音 · 计算机科学 2019-12-24 Ivo Trowitzsch , Christopher Schymura , Dorothea Kolossa , Klaus Obermayer

We present a new method to capture the acoustic characteristics of real-world rooms using commodity devices, and use the captured characteristics to generate similar sounding sources with virtual models. Given the captured audio and an…

声音 · 计算机科学 2021-09-28 Zhenyu Tang , Nicholas J. Bryan , Dingzeyu Li , Timothy R. Langlois , Dinesh Manocha

Whisper is one of the recent state-of-the-art multilingual speech recognition and translation models, however, it is not designed for real time transcription. In this paper, we build on top of Whisper and create Whisper-Streaming, an…

计算与语言 · 计算机科学 2023-09-22 Dominik Macháček , Raj Dabre , Ondřej Bojar

We present TagSpeech, a unified LLM-based framework that utilizes Temporal Anchor Grounding for joint multi-speaker ASR and diarization. The framework is built on two key designs: (1) decoupled semantic and speaker streams fine-tuned via…

音频与语音处理 · 电气工程与系统科学 2026-01-13 Mingyue Huo , Yiwen Shao , Yuheng Zhang

Speaker embeddings are promising identity-related features that can enhance the identity assignment performance of a tracking system by leveraging its spatial predictions, i.e, by performing identity reassignment. Common speaker embedding…

音频与语音处理 · 电气工程与系统科学 2025-08-21 Taous Iatariene , Alexandre Guérin , Romain Serizel

We propose a model of the speech perception of individual words in the presence of mishearings. This phenomenological approach is based on concepts used in linguistics, and provides a formalism that is universal across languages. We put…

计算与语言 · 计算机科学 2020-10-19 Anita Mehta , Jean-Marc Luck

Large general-purpose transformer models have recently become the mainstay in the realm of speech analysis. In particular, Whisper achieves state-of-the-art results in relevant tasks such as speech recognition, translation, language…

声音 · 计算机科学 2024-05-07 Antonio Bevilacqua , Paolo Saviano , Alessandro Amirante , Simon Pietro Romano

Spatial information is a critical clue for multi-channel multi-speaker target speech recognition. Most state-of-the-art multi-channel Automatic Speech Recognition (ASR) systems extract spatial features only during the speech separation…

音频与语音处理 · 电气工程与系统科学 2026-01-27 Yiwen Shao , Yong Xu , Sanjeev Khudanpur , Dong Yu

This paper describes the system developed by the XMUSPEECH team for the Multi-channel Multi-party Meeting Transcription Challenge (M2MeT). For the speaker diarization task, we propose a multi-channel speaker diarization system that obtains…

音频与语音处理 · 电气工程与系统科学 2022-02-14 Jie Wang , Yuji Liu , Binling Wang , Yiming Zhi , Song Li1 , Shipeng Xia , Jiayang Zhang , Lin Li1 , Qingyang Hong , Feng Tong