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This paper presents the details of the SRIB-LEAP submission to the ConferencingSpeech challenge 2021. The challenge involved the task of multi-channel speech enhancement to improve the quality of far field speech from microphone arrays in a…

音频与语音处理 · 电气工程与系统科学 2021-06-25 R G Prithvi Raj , Rohit Kumar , M K Jayesh , Anurenjan Purushothaman , Sriram Ganapathy , M A Basha Shaik

Single-channel speech enhancement is utilized in various tasks to mitigate the effect of interfering signals. Conventionally, to ensure the speech enhancement performs optimally, the speech enhancement has needed to be tuned for each task.…

音频与语音处理 · 电气工程与系统科学 2025-07-11 Hiroshi Sato , Tsubasa Ochiai , Marc Delcroix , Takafumi Moriya , Takanori Ashihara , Ryo Masumura

The deep learning based time-domain models, e.g. Conv-TasNet, have shown great potential in both single-channel and multi-channel speech enhancement. However, many experiments on the time-domain speech enhancement model are done in…

音频与语音处理 · 电气工程与系统科学 2021-10-28 Wangyou Zhang , Jing Shi , Chenda Li , Shinji Watanabe , Yanmin Qian

Purely neural network (NN) based speech separation and enhancement methods, although can achieve good objective scores, inevitably cause nonlinear speech distortions that are harmful for the automatic speech recognition (ASR). On the other…

音频与语音处理 · 电气工程与系统科学 2020-08-03 Yong Xu , Meng Yu , Shi-Xiong Zhang , Lianwu Chen , Chao Weng , Jianming Liu , Dong Yu

Multi-channel speech enhancement with ad-hoc sensors has been a challenging task. Speech model guided beamforming algorithms are able to recover natural sounding speech, but the speech models tend to be oversimplified or the inference would…

计算与语言 · 计算机科学 2018-02-16 Kaizhi Qian , Yang Zhang , Shiyu Chang , Xuesong Yang , Dinei Florencio , Mark Hasegawa-Johnson

Adapting pre-trained text Large Language Models (LLMs) into Speech Language Models (Speech LMs) via continual pretraining on speech data is promising, but often degrades the original text capabilities. We propose Multimodal Depth Upscaling,…

计算与语言 · 计算机科学 2026-04-02 Kazuki Yano , Jun Suzuki , Shinji Watanabe

So far, many of the deep learning approaches for voice conversion produce good quality speech by using a large amount of training data. This paper presents a Deep Bidirectional Long Short-Term Memory (DBLSTM) based voice conversion…

音频与语音处理 · 电气工程与系统科学 2018-09-27 Mingyang Zhang , Berrak Sisman , Sai Sirisha Rallabandi , Haizhou Li , Li Zhao

State-of-the-art neural network language models (NNLMs) represented by long short term memory recurrent neural networks (LSTM-RNNs) and Transformers are becoming highly complex. They are prone to overfitting and poor generalization when…

计算与语言 · 计算机科学 2022-08-30 Boyang Xue , Shoukang Hu , Junhao Xu , Mengzhe Geng , Xunying Liu , Helen Meng

Multimodal speech recognition aims to improve the performance of automatic speech recognition (ASR) systems by leveraging additional visual information that is usually associated to the audio input. While previous approaches make crucial…

声音 · 计算机科学 2022-04-29 Dan Oneata , Horia Cucu

Self-supervised learned models have been found to be very effective for certain speech tasks such as automatic speech recognition, speaker identification, keyword spotting and others. While the features are undeniably useful in speech…

音频与语音处理 · 电气工程与系统科学 2024-03-05 Ravi Shankar , Ke Tan , Buye Xu , Anurag Kumar

Although deep learning (DL) has achieved notable progress in speech enhancement (SE), further research is still required for a DL-based SE system to adapt effectively and efficiently to particular speakers. In this study, we propose a novel…

音频与语音处理 · 电气工程与系统科学 2021-11-11 Cheng Yu , Szu-Wei Fu , Tsun-An Hsieh , Yu Tsao , Mirco Ravanelli

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

Recently sequence-to-sequence models have started to achieve state-of-the-art performance on standard speech recognition tasks when processing audio data in batch mode, i.e., the complete audio data is available when starting processing.…

音频与语音处理 · 电气工程与系统科学 2020-07-28 Thai-Son Nguyen , Ngoc-Quan Pham , Sebastian Stueker , Alex Waibel

The use of spatial information with multiple microphones can improve far-field automatic speech recognition (ASR) accuracy. However, conventional microphone array techniques degrade speech enhancement performance when there is an array…

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

Recovering the masked speech frames is widely applied in speech representation learning. However, most of these models use random masking in the pre-training. In this work, we proposed two kinds of masking approaches: (1) speech-level…

声音 · 计算机科学 2022-10-26 Xulong Zhang , Jianzong Wang , Ning Cheng , Kexin Zhu , Jing Xiao

Recently, phase processing is attracting increasinginterest in speech enhancement community. Some researchersintegrate phase estimations module into speech enhancementmodels by using complex-valued short-time Fourier transform(STFT)…

声音 · 计算机科学 2019-01-03 Xingjian Du , Mengyao Zhu , Xuan Shi , Xinpeng Zhang , Wen Zhang , Jingdong Chen

This work analyzes how attention-based Bidirectional Long Short-Term Memory (BLSTM) models adapt to noise-augmented speech. We identify crucial components for noise adaptation in BLSTM models by freezing model components during fine-tuning.…

音频与语音处理 · 电气工程与系统科学 2021-09-27 Coleman Hooper , Thierry Tambe , Gu-Yeon Wei

To let the state-of-the-art end-to-end ASR model enjoy data efficiency, as well as much more unpaired text data by multi-modal training, one needs to address two problems: 1) the synchronicity of feature sampling rates between speech and…

音频与语音处理 · 电气工程与系统科学 2022-11-02 Yuhang Yang , Haihua Xu , Hao Huang , Eng Siong Chng , Sheng Li

The goal of this work is to develop a meeting transcription system that can recognize speech even when utterances of different speakers are overlapped. While speech overlaps have been regarded as a major obstacle in accurately transcribing…

音频与语音处理 · 电气工程与系统科学 2018-10-10 Takuya Yoshioka , Hakan Erdogan , Zhuo Chen , Xiong Xiao , Fil Alleva

While the transformer has emerged as the eminent neural architecture, several independent lines of research have emerged to address its limitations. Recurrent neural approaches have observed a lot of renewed interest, including the extended…

声音 · 计算机科学 2025-08-20 Sarthak Yadav , Sergios Theodoridis , Zheng-Hua Tan