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相关论文: Quantization for OpenAI's Whisper Models: A Compar…

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Model quantization has become essential for efficient large language model deployment, yet existing approaches involve clear trade-offs: methods such as GPTQ and AWQ achieve practical compression but are lossy, while lossless techniques…

机器学习 · 计算机科学 2026-05-05 Michael Helcig , Eldar Kurtic , Dan Alistarh

Discrete audio representations are gaining traction in speech modeling due to their interpretability and compatibility with large language models, but are not always optimized for noisy or real-world environments. Building on existing works…

计算与语言 · 计算机科学 2025-10-30 Shreyas Gopal , Ashutosh Anshul , Haoyang Li , Yue Heng Yeo , Hexin Liu , Eng Siong Chng

Recent research on word-level confidence estimation for speech recognition systems has primarily focused on lightweight models known as Confidence Estimation Modules (CEMs), which rely on hand-engineered features derived from Automatic…

音频与语音处理 · 电气工程与系统科学 2025-02-20 Vaibhav Aggarwal , Shabari S Nair , Yash Verma , Yash Jogi

This paper examines the integration of real-time talking-head generation for interviewer training, focusing on overcoming challenges in Audio Feature Extraction (AFE), which often introduces latency and limits responsiveness in real-time…

As Deep Neural Networks (DNNs) rapidly advance in various fields, including speech verification, they typically involve high computational costs and substantial memory consumption, which can be challenging to manage on mobile systems.…

音频与语音处理 · 电气工程与系统科学 2024-07-15 Yeona Hong , Woo-Jin Chung , Hong-Goo Kang

Reducing the latency and model size has always been a significant research problem for live Automatic Speech Recognition (ASR) application scenarios. Along this direction, model quantization has become an increasingly popular approach to…

音频与语音处理 · 电气工程与系统科学 2023-03-06 Shaojin Ding , Phoenix Meadowlark , Yanzhang He , Lukasz Lew , Shivani Agrawal , Oleg Rybakov

Natural language processing of conversational speech requires the availability of high-quality transcripts. In this paper, we express our skepticism towards the recent reports of very low Word Error Rates (WERs) achieved by modern Automatic…

Automatic Speech Recognition (ASR) systems often struggle with transcribing child speech due to the lack of large child speech datasets required to accurately train child-friendly ASR models. However, there are huge amounts of annotated…

音频与语音处理 · 电气工程与系统科学 2023-07-26 Rishabh Jain , Andrei Barcovschi , Mariam Yiwere , Peter Corcoran , Horia Cucu

This article concerns comparative studies on the Automatic Speech Recognition (ASR) model incorporated with the Large Language Model (LLM) used for medical interviews. The proposed solution is tested on polish language benchmarks and…

音频与语音处理 · 电气工程与系统科学 2026-03-04 Marcin Pietroń , Szymon Piórkowski , Kamil Faber , Dominik Żurek , Michał Karwatowski , Jerzy Duda , Hubert Zieliński , Piotr Lipnicki , Mikołaj Leszczuk

Most existing automatic speech recognition (ASR) research evaluate models using in-domain datasets. However, they seldom evaluate how they generalize across diverse speech contexts. This study addresses this gap by benchmarking seven Akan…

Automatic Speech Recognition systems have made significant progress with large-scale pre-trained models. However, most current systems focus solely on transcribing the speech without identifying speaker roles, a function that is critical…

音频与语音处理 · 电气工程与系统科学 2025-06-13 Anfeng Xu , Tiantian Feng , Shrikanth Narayanan

Automatic Speech Recognition (ASR) systems, such as Whisper, achieve high transcription accuracy but struggle with named entities and numerical data, especially when proper formatting is required. These issues increase word error rate (WER)…

计算与语言 · 计算机科学 2025-07-01 Duygu Altinok

As speech recognition model sizes and training data requirements grow, it is increasingly common for systems to only be available via APIs from online service providers rather than having direct access to models themselves. In this scenario…

音频与语音处理 · 电气工程与系统科学 2023-10-11 Rao Ma , Mengjie Qian , Mark J. F. Gales , Kate M. Knill

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

This study explores fine-tuning multilingual ASR (Automatic Speech Recognition) models, specifically OpenAI's Whisper-Tiny, to improve performance in Japanese. While multilingual models like Whisper offer versatility, they often lack…

计算与语言 · 计算机科学 2024-12-17 Mark Bajo , Haruka Fukukawa , Ryuji Morita , Yuma Ogasawara

We demonstrate that carefully adjusting the tokenizer of the Whisper speech recognition model significantly improves the precision of word-level timestamps when applying dynamic time warping to the decoder's cross-attention scores. We…

机器学习 · 计算机科学 2024-08-30 Laurin Wagner , Bernhard Thallinger , Mario Zusag

For on-device automatic speech recognition (ASR), quantization aware training (QAT) is ubiquitous to achieve the trade-off between model predictive performance and efficiency. Among existing QAT methods, one major drawback is that the…

The Whisper model, an open-source automatic speech recognition system, is widely adopted for its strong performance across multilingual and zero-shot settings. However, it frequently suffers from hallucination errors, especially under noisy…

人工智能 · 计算机科学 2025-11-19 Kumud Tripathi , Aditya Srinivas Menon , Aman Gaurav , Raj Prakash Gohil , Pankaj Wasnik

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

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