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相关论文: Non causal deep learning based dereverberation

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The attention-based deep contextual biasing method has been demonstrated to effectively improve the recognition performance of end-to-end automatic speech recognition (ASR) systems on given contextual phrases. However, unlike shallow fusion…

音频与语音处理 · 电气工程与系统科学 2023-10-10 Kaixun Huang , Ao Zhang , Binbin Zhang , Tianyi Xu , Xingchen Song , Lei Xie

Automatic speech recognition (ASR) systems often make unrecoverable errors due to subsystem pruning (acoustic, language and pronunciation models); for example pruning words due to acoustics using short-term context, prior to rescoring with…

计算与语言 · 计算机科学 2019-07-01 Prashanth Gurunath Shivakumar , Haoqi Li , Kevin Knight , Panayiotis Georgiou

The performances of automatic speech recognition (ASR) systems are usually evaluated by the metric word error rate (WER) when the manually transcribed data are provided, which are, however, expensively available in the real scenario. In…

计算与语言 · 计算机科学 2020-09-01 Kai Fan , Jiayi Wang , Bo Li , Shiliang Zhang , Boxing Chen , Niyu Ge , Zhijie Yan

Contextual ASR, which takes a list of bias terms as input along with audio, has drawn recent interest as ASR use becomes more widespread. We are releasing contextual biasing lists to accompany the Earnings21 dataset, creating a public…

计算与语言 · 计算机科学 2022-09-07 Jennifer Drexler Fox , Natalie Delworth

A promising approach for speech dereverberation is based on supervised learning, where a deep neural network (DNN) is trained to predict the direct sound from noisy-reverberant speech. This data-driven approach is based on leveraging prior…

声音 · 计算机科学 2021-11-11 Zhong-Qiu Wang , Gordon Wichern , Jonathan Le Roux

Environmental noises and reverberation have a detrimental effect on the performance of automatic speech recognition (ASR) systems. Multi-condition training of neural network-based acoustic models is used to deal with this problem, but it…

音频与语音处理 · 电气工程与系统科学 2021-02-03 Desh Raj , Jesus Villalba , Daniel Povey , Sanjeev Khudanpur

The common standard for quality evaluation of automatic speech recognition (ASR) systems is reference-based metrics such as the Word Error Rate (WER), computed using manual ground-truth transcriptions that are time-consuming and expensive…

计算与语言 · 计算机科学 2023-06-26 Kamer Ali Yuksel , Thiago Ferreira , Ahmet Gunduz , Mohamed Al-Badrashiny , Golara Javadi

This paper addresses the challenge of integrating low-resource languages into multilingual automatic speech recognition (ASR) systems. We introduce a novel application of weighted cross-entropy, typically used for unbalanced datasets, to…

Neural networks (NNs) have been widely applied in speech processing tasks, and, in particular, those employing microphone arrays. Nevertheless, most existing NN architectures can only deal with fixed and position-specific microphone arrays.…

音频与语音处理 · 电气工程与系统科学 2021-06-14 Yochai Yemini , Ethan Fetaya , Haggai Maron , Sharon Gannot

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 method for multi-channel time domain speech separation in reverberant environments. A fully-convolutional neural network structure has been used to directly separate speech from multiple microphone recordings,…

音频与语音处理 · 电气工程与系统科学 2020-11-12 Jisi Zhang , Catalin Zorila , Rama Doddipatla , Jon Barker

In the realm of automatic speech recognition (ASR), the quest for models that not only perform with high accuracy but also offer transparency in their decision-making processes is crucial. The potential of quality estimation (QE) metrics is…

计算与语言 · 计算机科学 2024-02-06 Golara Javadi , Kamer Ali Yuksel , Yunsu Kim , Thiago Castro Ferreira , Mohamed Al-Badrashiny

We propose new, data-efficient training tasks for BERT models that improve performance of automatic speech recognition (ASR) systems on conversational speech. We include past conversational context and fine-tune BERT on transcript…

计算与语言 · 计算机科学 2022-01-26 Pablo Ortiz , Simen Burud

Recent advancement in deep learning encouraged developing large automatic speech recognition (ASR) models that achieve promising results while ignoring computational and memory constraints. However, deploying such models on low resource…

计算机视觉与模式识别 · 计算机科学 2025-05-29 Abdul Hannan , Alessio Brutti , Shah Nawaz , Mubashir Noman

Reverberations are unavoidable in enclosures, resulting in reduced intelligibility for hearing impaired and non native listeners and even for the normal hearing listeners in noisy circumstances. It also degrades the performance of machine…

音频与语音处理 · 电气工程与系统科学 2022-08-11 Sania Gul , Muhammad Salman Khan , Syed Waqar Shah

Automatic Speech Recognition (ASR) systems suffer considerably when source speech is corrupted with noise or room impulse responses (RIR). Typically, speech enhancement is applied in both mismatched and matched scenario training and…

音频与语音处理 · 电气工程与系统科学 2022-04-26 Shashi Kumar , Shakti P. Rath , Abhishek Pandey

Building an accurate automatic speech recognition (ASR) system requires a large dataset that contains many hours of labeled speech samples produced by a diverse set of speakers. The lack of such open free datasets is one of the main issues…

计算与语言 · 计算机科学 2018-11-05 Jason Li , Ravi Gadde , Boris Ginsburg , Vitaly Lavrukhin

Sequence-to-sequence models, such as attention-based models in automatic speech recognition (ASR), are typically trained to optimize the cross-entropy criterion which corresponds to improving the log-likelihood of the data. However, system…

计算与语言 · 计算机科学 2017-12-06 Rohit Prabhavalkar , Tara N. Sainath , Yonghui Wu , Patrick Nguyen , Zhifeng Chen , Chung-Cheng Chiu , Anjuli Kannan

Developing a single-microphone speech denoising or dereverberation front-end for robust automatic speaker verification (ASV) in noisy far-field speaking scenarios is challenging. To address this problem, we present a novel front-end design…

音频与语音处理 · 电气工程与系统科学 2021-12-28 Joon-Young Yang , Joon-Hyuk Chang

We present a novel multi-channel front-end based on channel shortening with theWeighted Prediction Error (WPE) method followed by a fixed MVDR beamformer used in combination with a recently proposed self-attention-based channel combination…

音频与语音处理 · 电气工程与系统科学 2022-03-29 Dushyant Sharma , Rong Gong , James Fosburgh , Stanislav Yu. Kruchinin , Patrick A. Naylor , Ljubomir Milanovic