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Machine recognition of an atypical speech like whispered speech, is a challenging task. We introduce whisper-to-natural-speech conversion using sequence-to-sequence approach by proposing enhanced transformer architecture, which uses both…

音频与语音处理 · 电气工程与系统科学 2021-04-06 Abhishek Niranjan , Mukesh Sharma , Sai Bharath Chandra Gutha , M Ali Basha Shaik

Recent advancements in multilingual automatic speech recognition (ASR) have been driven by large-scale end-to-end models like Whisper. However, challenges such as language interference and expanding to unseen languages (language expansion)…

计算与语言 · 计算机科学 2025-09-29 Hongli Yang , Sheng Li , Hao Huang , Ayiduosi Tuohan , Yizhou Peng

Recent techniques for speech deepfake detection often rely on pre-trained self-supervised models. These systems, initially developed for Automatic Speech Recognition (ASR), have proved their ability to offer a meaningful representation of…

Spectrum sensing is an essential component of modern wireless networks as it offers a tool to characterize spectrum usage and better utilize it. Deep Learning (DL) has become one of the most used techniques to perform spectrum sensing as…

网络与互联网体系结构 · 计算机科学 2024-01-11 Clifton Paul Robinson , Daniel Uvaydov , Salvatore D'Oro , Tommaso Melodia

Self-supervised speech representation learning (SSL) has shown to be effective in various downstream tasks, but SSL models are usually large and slow. Model compression techniques such as pruning aim to reduce the model size and computation…

计算与语言 · 计算机科学 2023-03-01 Yifan Peng , Kwangyoun Kim , Felix Wu , Prashant Sridhar , Shinji Watanabe

Transformers and their variants have achieved great success in speech processing. However, their multi-head self-attention mechanism is computationally expensive. Therefore, one novel selective state space model, Mamba, has been proposed as…

音频与语音处理 · 电气工程与系统科学 2025-03-04 Yang Xiao , Rohan Kumar Das

This paper introduces WaveNet, a deep neural network for generating raw audio waveforms. The model is fully probabilistic and autoregressive, with the predictive distribution for each audio sample conditioned on all previous ones;…

This paper presents a novel high-fidelity and low-latency universal neural vocoder framework based on multiband WaveRNN with data-driven linear prediction for discrete waveform modeling (MWDLP). MWDLP employs a coarse-fine bit WaveRNN…

声音 · 计算机科学 2021-07-06 Patrick Lumban Tobing , Tomoki Toda

Wireless signal recognition (WSR) is crucial in modern and future wireless communication networks since it aims to identify properties of the received signal. Although many deep learning-based WSR models have been developed, they still rely…

信号处理 · 电气工程与系统科学 2024-04-04 Hao Zhang , Fuhui Zhou , Qihui Wu , Naofal Al-Dhahir

The convolutional neural network (CNN) based approaches have shown great success for speaker verification (SV) tasks, where modeling long temporal context and reducing information loss of speaker characteristics are two important challenges…

声音 · 计算机科学 2021-08-31 Yanfeng Wu , Chenkai Guo , Junan Zhao , Xiao Jin , Jing Xu

The rapid evolution of deepfake generation technologies necessitates the development of robust face forgery detection algorithms. Recent studies have demonstrated that wavelet analysis can enhance the generalization abilities of forgery…

计算机视觉与模式识别 · 计算机科学 2025-10-22 Siran Peng , Tianshuo Zhang , Li Gao , Xiangyu Zhu , Haoyuan Zhang , Kai Pang , Zhen Lei

Conventional spoofing detection systems have heavily relied on the use of handcrafted features derived from speech data. However, a notable shift has recently emerged towards the direct utilization of raw speech waveforms, as demonstrated…

Data augmentation is conventionally used to inject robustness in Speaker Verification systems. Several recently organized challenges focus on handling novel acoustic environments. Deep learning based speech enhancement is a modern solution…

音频与语音处理 · 电气工程与系统科学 2020-04-29 Saurabh Kataria , Phani Sankar Nidadavolu , Jesús Villalba , Najim Dehak

With excellent generalization ability, self-supervised speech models have shown impressive performance on various downstream speech tasks in the pre-training and fine-tuning paradigm. However, as the growing size of pre-trained models,…

音频与语音处理 · 电气工程与系统科学 2024-03-04 Mufan Sang , John H. L. Hansen

This paper presents XLSR which learns cross-lingual speech representations by pretraining a single model from the raw waveform of speech in multiple languages. We build on wav2vec 2.0 which is trained by solving a contrastive task over…

计算与语言 · 计算机科学 2020-12-17 Alexis Conneau , Alexei Baevski , Ronan Collobert , Abdelrahman Mohamed , Michael Auli

This paper describes our submitted systems to the ASVspoof 5 Challenge Track 1: Speech Deepfake Detection - Open Condition, which consists of a stand-alone speech deepfake (bonafide vs spoof) detection task. Recently, large-scale…

音频与语音处理 · 电气工程与系统科学 2025-06-25 Theophile Stourbe , Victor Miara , Theo Lepage , Reda Dehak

Self-supervised speech representation learning (speech SSL) has demonstrated the benefit of scale in learning rich representations for Automatic Speech Recognition (ASR) with limited paired data, such as wav2vec 2.0. We investigate the…

In this paper, we propose a progressive learning paradigm for transformer-based variable-rate image compression. Our approach covers a wide range of compression rates with the assistance of the Layer-adaptive Prompt Module (LPM). Inspired…

计算机视觉与模式识别 · 计算机科学 2023-11-29 Shiyu Qin , Yimin Zhou , Jinpeng Wang , Bin Chen , Baoyi An , Tao Dai , Shu-Tao Xia

The increasing of digital radio frequency memory based electronic countermeasures poses a significant threat to the survivability and effectiveness of radar systems. These jammers can generate a multitude of deceptive false targets,…

信号处理 · 电气工程与系统科学 2025-10-22 Yizhen Jia , Siyao Xiao , Wenkai Jia , Hui Chen , Wen-Qin Wang

Recent advances in artificial speech and audio technologies have improved the abilities of deep-fake operators to falsify media and spread malicious misinformation. Anyone with limited coding skills can use freely available speech synthesis…