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There is often a trade-off between performance and latency in streaming automatic speech recognition (ASR). Traditional methods such as look-ahead and chunk-based methods, usually require information from future frames to advance…

音频与语音处理 · 电气工程与系统科学 2022-07-07 Zehan Li , Haoran Miao , Keqi Deng , Gaofeng Cheng , Sanli Tian , Ta Li , Yonghong Yan

In this paper, we present Adaptive Computation Steps (ACS) algo-rithm, which enables end-to-end speech recognition models to dy-namically decide how many frames should be processed to predict a linguistic output. The model that applies ACS…

音频与语音处理 · 电气工程与系统科学 2018-09-27 Mohan Li , Min Liu , Masanori Hattori

Modern systems for automatic speech recognition, including the RNN-Transducer and Attention-based Encoder-Decoder (AED), are designed so that the encoder is not required to alter the time-position of information from the audio sequence into…

声音 · 计算机科学 2025-02-11 Adam Stooke , Rohit Prabhavalkar , Khe Chai Sim , Pedro Moreno Mengibar

Non-autoregressive automatic speech recognition (ASR) has become a mainstream of ASR modeling because of its fast decoding speed and satisfactory result. To further boost the performance, relaxing the conditional independence assumption and…

计算与语言 · 计算机科学 2023-05-19 Chong-En Lin , Kuan-Yu Chen

Conventional far-field automatic speech recognition (ASR) systems typically employ microphone array techniques for speech enhancement in order to improve robustness against noise or reverberation. However, such speech enhancement techniques…

音频与语音处理 · 电气工程与系统科学 2021-12-23 Minhua Wu , Kenichi Kumatani , Shiva Sundaram , Nikko Strom , Bjorn Hoffmeister

We propose an unsupervised speaker adaptation method inspired by the neural Turing machine for end-to-end (E2E) automatic speech recognition (ASR). The proposed model contains a memory block that holds speaker i-vectors extracted from the…

音频与语音处理 · 电气工程与系统科学 2020-02-17 Leda Sarı , Niko Moritz , Takaaki Hori , Jonathan Le Roux

Audio-visual speech enhancement system is regarded to be one of promising solutions for isolating and enhancing speech of desired speaker. Conventional methods focus on predicting clean speech spectrum via a naive convolution neural network…

音频与语音处理 · 电气工程与系统科学 2022-09-28 Xinmeng Xu , Jianjun Hao

While linear attention reduces the quadratic complexity of standard Transformers to linear time, it often lags behind in expressivity due to the removal of softmax normalization. This omission eliminates \emph{global competition}, a…

机器学习 · 计算机科学 2026-02-03 Mingwei Xu , Xuan Lin , Xinnan Guo , Wanqing Xu , Wanyun Cui

In interactive automatic speech recognition (ASR) systems, low-latency requirements limit the amount of search space that can be explored during decoding, particularly in end-to-end neural ASR. In this paper, we present a novel streaming…

音频与语音处理 · 电气工程与系统科学 2024-01-29 Denis Filimonov , Prabhat Pandey , Ariya Rastrow , Ankur Gandhe , Andreas Stolcke

As end-to-end automatic speech recognition (ASR) models reach promising performance, various downstream tasks rely on good confidence estimators for these systems. Recent research has shown that model-based confidence estimators have a…

音频与语音处理 · 电气工程与系统科学 2022-03-03 Qiujia Li , Yu Zhang , David Qiu , Yanzhang He , Liangliang Cao , Philip C. Woodland

While end-to-end (E2E) automatic speech recognition (ASR) models excel at general transcription, they struggle to recognize rare or unseen named entities (e.g., contact names, locations), which are critical for downstream applications like…

计算与语言 · 计算机科学 2025-12-22 Ramaneswaran Selvakumar , Cindy Tseng , Eesung Kim , Vijendra Raj Apsingekar , Yun Tang

We address the fundamental incompatibility of attention-based encoder-decoder (AED) models with long-form acoustic encodings. AED models trained on segmented utterances learn to encode absolute frame positions by exploiting limited acoustic…

音频与语音处理 · 电气工程与系统科学 2025-12-17 Pawel Swietojanski , Xinwei Li , Mingbin Xu , Takaaki Hori , Dogan Can , Xiaodan Zhuang

This paper presents a novel study of parameter-free attentive scoring for speaker verification. Parameter-free scoring provides the flexibility of comparing speaker representations without the need of an accompanying parametric scoring…

声音 · 计算机科学 2023-03-07 Jason Pelecanos , Quan Wang , Yiling Huang , Ignacio Lopez Moreno

Attention-based contextual biasing approaches have shown significant improvements in the recognition of generic and/or personal rare-words in End-to-End Automatic Speech Recognition (E2E ASR) systems like neural transducers. These…

This study addresses robust automatic speech recognition (ASR) by introducing a Conformer-based acoustic model. The proposed model builds on the wide residual bi-directional long short-term memory network (WRBN) with utterance-wise dropout…

声音 · 计算机科学 2022-10-21 Yufeng Yang , Peidong Wang , DeLiang Wang

We propose a cross-modal transformer-based neural correction models that refines the output of an automatic speech recognition (ASR) system so as to exclude ASR errors. Generally, neural correction models are composed of encoder-decoder…

Training large foundation models using self-supervised objectives on unlabeled data, followed by fine-tuning on downstream tasks, has emerged as a standard procedure. Unfortunately, the efficacy of this approach is often constrained by both…

Recurrent sequence generators conditioned on input data through an attention mechanism have recently shown very good performance on a range of tasks in- cluding machine translation, handwriting synthesis and image caption gen- eration. We…

计算与语言 · 计算机科学 2015-06-25 Jan Chorowski , Dzmitry Bahdanau , Dmitriy Serdyuk , Kyunghyun Cho , Yoshua Bengio

Automatic speech recognition (ASR) models with low-footprint are increasingly being deployed on edge devices for conversational agents, which enhances privacy. We study the problem of federated continual incremental learning for recurrent…

音频与语音处理 · 电气工程与系统科学 2023-06-22 Milind Rao , Gopinath Chennupati , Gautam Tiwari , Anit Kumar Sahu , Anirudh Raju , Ariya Rastrow , Jasha Droppo

This work introduces the Cleanformer, a streaming multichannel neural based enhancement frontend for automatic speech recognition (ASR). This model has a conformer-based architecture which takes as inputs a single channel each of raw and…

音频与语音处理 · 电气工程与系统科学 2023-05-05 Joseph Caroselli , Arun Narayanan , Nathan Howard , Tom O'Malley