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相关论文: RIR-Mega-Speech: A Reverberant Speech Corpus with …

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We introduce Whisper-RIR-Mega, a benchmark dataset of paired clean and reverberant speech for evaluating automatic speech recognition (ASR) robustness to room acoustics. Each sample pairs a clean LibriSpeech utterance with the same…

音频与语音处理 · 电气工程与系统科学 2026-03-17 Mandip Goswami

Room impulse responses are a core resource for dereverberation, robust speech recognition, source localization, and room acoustics estimation. We present RIR-Mega, a large collection of simulated RIRs described by a compact, machine…

音频与语音处理 · 电气工程与系统科学 2025-10-29 Mandip Goswami

This paper introduces GigaSpeech, an evolving, multi-domain English speech recognition corpus with 10,000 hours of high quality labeled audio suitable for supervised training, and 40,000 hours of total audio suitable for semi-supervised and…

This paper presents BUT ReverbDB - a dataset of real room impulse responses (RIR), background noises and re-transmitted speech data. The retransmitted data includes LibriSpeech test-clean, 2000 HUB5 English evaluation and part of 2010 NIST…

音频与语音处理 · 电气工程与系统科学 2019-09-04 Igor Szoke , Miroslav Skacel , Ladislav Mosner , Jakub Paliesek , Jan "Honza" Cernocky

We present a novel approach that improves the performance of reverberant speech separation. Our approach is based on an accurate geometric acoustic simulator (GAS) which generates realistic room impulse responses (RIRs) by modeling both…

音频与语音处理 · 电气工程与系统科学 2021-07-21 Rohith Aralikatti , Anton Ratnarajah , Zhenyu Tang , Dinesh Manocha

This paper addresses the combination of complementary parallel speech recognition systems to reduce the error rate of speech recognition systems operating in real highly-reverberant environments. First, the testing environment consists of…

音频与语音处理 · 电气工程与系统科学 2019-06-19 José Novoa , Josué Fredes , Jorge Wuth , Fernando Huenupán , Richard M. Stern , Nestor Becerra Yoma

This paper introduces a new speech dataset called ``LibriTTS-R'' designed for text-to-speech (TTS) use. It is derived by applying speech restoration to the LibriTTS corpus, which consists of 585 hours of speech data at 24 kHz sampling rate…

音频与语音处理 · 电气工程与系统科学 2023-05-31 Yuma Koizumi , Heiga Zen , Shigeki Karita , Yifan Ding , Kohei Yatabe , Nobuyuki Morioka , Michiel Bacchiani , Yu Zhang , Wei Han , Ankur Bapna

Speech restoration (SR) is a task of converting degraded speech signals into high-quality ones. In this study, we propose a robust SR model called Miipher, and apply Miipher to a new SR application: increasing the amount of high-quality…

Blind estimation of acoustic room parameters such as the reverberation time $T_\mathrm{60}$ and the direct-to-reverberation ratio ($\mathrm{DRR}$) is still a challenging task, especially in case of blind estimation from reverberant speech…

声音 · 计算机科学 2015-10-16 Feifei Xiong , Stefan Goetze , Bernd T. Meyer

Noise robustness is essential for deploying automatic speech recognition (ASR) systems in real-world environments. One way to reduce the effect of noise interference is to employ a preprocessing module that conducts speech enhancement, and…

High quality transcription data is crucial for training automatic speech recognition (ASR) systems. However, the existing industry-level data collection pipelines are expensive to researchers, while the quality of crowdsourced transcription…

计算与语言 · 计算机科学 2023-09-27 Jian Gao , Hanbo Sun , Cheng Cao , Zheng Du

The incompleteness of speech inputs severely degrades the performance of all the related speech signal processing applications. Although many researches have been proposed to address this issue, they controlled the data missing conditions…

声音 · 计算机科学 2022-01-25 Yu-Chen Lin , Tsun-An Hsieh , Kuo-Hsuan Hung , Cheng Yu , Harinath Garudadri , Yu Tsao , Tei-Wei Kuo

For conversational large-vocabulary continuous speech recognition (LVCSR) tasks, up to about two thousand hours of audio is commonly used to train state of the art models. Collection of labeled conversational audio however, is prohibitively…

计算与语言 · 计算机科学 2017-05-30 Shane Walker , Morten Pedersen , Iroro Orife , Jason Flaks

Automatic speech recognition (ASR) performs well for high-resource languages with abundant paired audio-transcript data, but its accuracy degrades sharply for most languages due to limited publicly available aligned data. To this end, we…

计算与语言 · 计算机科学 2026-05-12 Antonis Asonitis , Luca A. Lanzendörfer , Frédéric Berdoz , Roger Wattenhofer

With the development of deep learning, automatic speech recognition (ASR) has made significant progress. To further enhance the performance of ASR, revising recognition results is one of the lightweight but efficient manners. Various…

计算与语言 · 计算机科学 2024-06-14 Yi-Wei Wang , Ke-Han Lu , Kuan-Yu Chen

This paper introduces a new speech corpus called "LibriTTS" designed for text-to-speech use. It is derived from the original audio and text materials of the LibriSpeech corpus, which has been used for training and evaluating automatic…

声音 · 计算机科学 2019-04-08 Heiga Zen , Viet Dang , Rob Clark , Yu Zhang , Ron J. Weiss , Ye Jia , Zhifeng Chen , Yonghui Wu

It is well known that many machine learning systems demonstrate bias towards specific groups of individuals. This problem has been studied extensively in the Facial Recognition area, but much less so in Automatic Speech Recognition (ASR).…

音频与语音处理 · 电气工程与系统科学 2021-11-22 Chunxi Liu , Michael Picheny , Leda Sarı , Pooja Chitkara , Alex Xiao , Xiaohui Zhang , Mark Chou , Andres Alvarado , Caner Hazirbas , Yatharth Saraf

Speech recognition in highly-reverberant real environments remains a major challenge. An evaluation dataset for this task is needed. This report describes the generation of the Highly-Reverberant Real Environment database (HRRE). This…

音频与语音处理 · 电气工程与系统科学 2018-03-28 Juan Pablo Escudero , Victor Poblete , José Novoa , Jorge Wuth , Josué Fredes , Rodrigo Mahu , Richard Stern , Néstor Becerra Yoma

This paper proposes a speech enhancement method which exploits the high potential of residual connections in a Wide Residual Network architecture. This is supported on single dimensional convolutions computed alongside the time domain,…

音频与语音处理 · 电气工程与系统科学 2019-04-11 Jorge Llombart , Dayana Ribas , Antonio Miguel , Luis Vicente , Alfonso Ortega , Eduardo Lleida

Text encodings from automatic speech recognition (ASR) transcripts and audio representations have shown promise in speech emotion recognition (SER) ever since. Yet, it is challenging to explain the effect of each information stream on the…

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