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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

There is a wide variety of speech processing tasks ranging from extracting content information from speech signals to generating speech signals. For different tasks, model networks are usually designed and tuned separately. If a universal…

In recent years, various flow-based generative models have been proposed to generate high-fidelity waveforms in real-time. However, these models require either a well-trained teacher network or a number of flow steps making them…

声音 · 计算机科学 2020-07-06 Hyeongju Kim , Hyeonseung Lee , Woo Hyun Kang , Sung Jun Cheon , Byoung Jin Choi , Nam Soo Kim

Neural waveform models such as the WaveNet are used in many recent text-to-speech systems, but the original WaveNet is quite slow in waveform generation because of its autoregressive (AR) structure. Although faster non-AR models were…

音频与语音处理 · 电气工程与系统科学 2019-04-30 Xin Wang , Shinji Takaki , Junichi Yamagishi

We propose a novel decentralized feature extraction approach in federated learning to address privacy-preservation issues for speech recognition. It is built upon a quantum convolutional neural network (QCNN) composed of a quantum circuit…

Distributed microphone array (DMA) is a promising next-generation platform for speech interaction, where speech enhancement (SE) is still required to improve the speech quality in noisy cases. Existing SE methods usually first gather raw…

声音 · 计算机科学 2026-01-27 Chengqian Jiang , Jie Zhang , Haoyin Yan

Compensation for channel mismatch and noise interference is essential for robust automatic speech recognition. Enhanced speech has been introduced into the multi-condition training of acoustic models to improve their generalization ability.…

声音 · 计算机科学 2022-11-24 Hung-Shin Lee , Pin-Yuan Chen , Yao-Fei Cheng , Yu Tsao , Hsin-Min Wang

Most deep learning-based models for speech enhancement have mainly focused on estimating the magnitude of spectrogram while reusing the phase from noisy speech for reconstruction. This is due to the difficulty of estimating the phase of…

声音 · 计算机科学 2019-04-03 Hyeong-Seok Choi , Jang-Hyun Kim , Jaesung Huh , Adrian Kim , Jung-Woo Ha , Kyogu Lee

There are many deterministic mathematical operations (e.g. compression, clipping, downsampling) that degrade speech quality considerably. In this paper we introduce a neural network architecture, based on a modification of the DiffWave…

声音 · 计算机科学 2021-09-03 Jianwei Zhang , Suren Jayasuriya , Visar Berisha

Robust speech processing in multi-talker environments requires effective speech separation. Recent deep learning systems have made significant progress toward solving this problem, yet it remains challenging particularly in real-time, short…

声音 · 计算机科学 2018-04-19 Yi Luo , Nima Mesgarani

One key step in audio signal processing is to transform the raw signal into representations that are efficient for encoding the original information. Traditionally, people transform the audio into spectral representations, as a function of…

声音 · 计算机科学 2016-11-30 Shuhui Qu , Juncheng Li , Wei Dai , Samarjit Das

In the past few years, convolutional neural networks (CNNs) have achieved milestones in medical image analysis. Especially, the deep neural networks based on U-shaped architecture and skip-connections have been widely applied in a variety…

图像与视频处理 · 电气工程与系统科学 2021-05-13 Hu Cao , Yueyue Wang , Joy Chen , Dongsheng Jiang , Xiaopeng Zhang , Qi Tian , Manning Wang

Wearable electrocardiogram (ECG) measurement using dry electrodes has a problem with high-intensity noise distortion. Hence, a robust noise reduction method is required. However, overlapping frequency bands of ECG and noise make noise…

信号处理 · 电气工程与系统科学 2025-01-14 Takamasa Terada , Masahiro Toyoura

The semi-airborne transient electromagnetic method (SATEM) is capable of conducting rapid surveys over large-scale and hard-to-reach areas. However, the acquired signals are often contaminated by complex noise, which can compromise the…

机器学习 · 计算机科学 2025-03-31 Shuang Wang , Ming Guo , Xuben Wang , Fei Deng , Lifeng Mao , Bin Wang , Wenlong Gao

Contemporary speech enhancement predominantly relies on audio transforms that are trained to reconstruct a clean speech waveform. The development of high-performing neural network sound recognition systems has raised the possibility of…

音频与语音处理 · 电气工程与系统科学 2025-11-18 Mark R. Saddler , Andrew Francl , Jenelle Feather , Kaizhi Qian , Yang Zhang , Josh H. McDermott

State-of-the-art singing voice separation is based on deep learning making use of CNN structures with skip connections (like U-net model, Wave-U-Net model, or MSDENSELSTM). A key to the success of these models is the availability of a large…

声音 · 计算机科学 2019-06-25 Alice Cohen-Hadria , Axel Roebel , Geoffroy Peeters

In this work, we extend ClariNet (Ping et al., 2019), a fully end-to-end speech synthesis model (i.e., text-to-wave), to generate high-fidelity speech from multiple speakers. To model the unique characteristic of different voices, low…

计算与语言 · 计算机科学 2019-07-11 Jihyun Park , Kexin Zhao , Kainan Peng , Wei Ping

Wavelets have emerged as a cutting edge technology in a number of fields. Concrete results of their application in Image and Signal processing suggest that wavelets can be effectively applied to Natural Language Processing (NLP) tasks that…

计算与语言 · 计算机科学 2025-08-04 Rana Salama , Abdou Youssef , Mona Diab

Recently, we made available WeNet, a production-oriented end-to-end speech recognition toolkit, which introduces a unified two-pass (U2) framework and a built-in runtime to address the streaming and non-streaming decoding modes in a single…

声音 · 计算机科学 2022-07-06 Binbin Zhang , Di Wu , Zhendong Peng , Xingchen Song , Zhuoyuan Yao , Hang Lv , Lei Xie , Chao Yang , Fuping Pan , Jianwei Niu

The voice mode of the Opus audio coder can compress wideband speech at bit rates ranging from 6 kb/s to 40 kb/s. However, Opus is at its core a waveform matching coder, and as the rate drops below 10 kb/s, quality degrades quickly. As the…

音频与语音处理 · 电气工程与系统科学 2020-08-11 Jan Skoglund , Jean-Marc Valin