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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

Unsupervised automatic speech recognition (ASR) aims to learn the mapping between the speech signal and its corresponding textual transcription without the supervision of paired speech-text data. A word/phoneme in the speech signal is…

音频与语音处理 · 电气工程与系统科学 2024-11-18 Liang-Hsuan Tseng , En-Pei Hu , Cheng-Han Chiang , Yuan Tseng , Hung-yi Lee , Lin-shan Lee , Shao-Hua Sun

In this paper, a speech enhancement method based on noise compensation performed on short time magnitude as well phase spectra is presented. Unlike the conventional geometric approach (GA) to spectral subtraction (SS), here the noise…

音频与语音处理 · 电气工程与系统科学 2018-03-09 Md Tauhidul Islam , Udoy Saha , K. T. Shahid , Ahmed Bin Hussain , Celia Shahnaz

We describe a modulation-domain loss function for deep-learning-based speech enhancement systems. Learnable spectro-temporal receptive fields (STRFs) were adapted to optimize for a speaker identification task. The learned STRFs were then…

音频与语音处理 · 电气工程与系统科学 2021-02-16 Tyler Vuong , Yangyang Xia , Richard M. Stern

The automatic identification and analysis of pronunciation errors, known as Mispronunciation Detection and Diagnosis (MDD) plays a crucial role in Computer Aided Pronunciation Learning (CAPL) tools such as Second-Language (L2) learning or…

音频与语音处理 · 电气工程与系统科学 2023-11-14 Mostafa Shahin , Julien Epps , Beena Ahmed

In recent decades, many studies have suggested that phase information is crucial for speech enhancement (SE), and time-domain single-channel speech enhancement techniques have shown promise in noise suppression and robust automatic speech…

音频与语音处理 · 电气工程与系统科学 2021-08-27 Fu-An Chao , Jeih-weih Hung , Berlin Chen

Traditional Automatic Video Dubbing (AVD) pipeline consists of three key modules, namely, Automatic Speech Recognition (ASR), Neural Machine Translation (NMT), and Text-to-Speech (TTS). Within AVD pipelines, isometric-NMT algorithms are…

While deep learning based speech enhancement systems have made rapid progress in improving the quality of speech signals, they can still produce outputs that contain artifacts and can sound unnatural. We propose a novel approach to speech…

声音 · 计算机科学 2022-07-12 Muqiao Yang , Joseph Konan , David Bick , Anurag Kumar , Shinji Watanabe , Bhiksha Raj

In this paper we present a single-microphone speech enhancement algorithm. A hybrid approach is proposed merging the generative mixture of Gaussians (MoG) model and the discriminative neural network (NN). The proposed algorithm is executed…

声音 · 计算机科学 2015-10-27 Shlomo E. Chazan , Jacob Goldberger , Sharon Gannot

We present a single-channel phase-sensitive speech enhancement algorithm that is based on modulation-domain Kalman filtering and on tracking the speech phase using circular statistics. With Kalman filtering, using that speech and noise are…

声音 · 计算机科学 2017-08-08 Nikolaos Dionelis , Mike Brookes

Phonemic segmentation of speech is a critical step of speech recognition systems. We propose a novel unsupervised algorithm based on sequence prediction models such as Markov chains and recurrent neural network. Our approach consists in…

计算与语言 · 计算机科学 2017-05-30 Paul Michel , Okko Räsänen , Roland Thiollière , Emmanuel Dupoux

Form about four decades human beings have been dreaming of an intelligent machine which can master the natural speech. In its simplest form, this machine should consist of two subsystems, namely automatic speech recognition (ASR) and speech…

声音 · 计算机科学 2013-05-08 Urmila Shrawankar , V. M. Thakare

So far, several physical models have been proposed for the study of vocal fold oscillations during phonation. The parameters of these models, such as vocal fold elasticity, resistance, etc. are traditionally determined through the…

声音 · 计算机科学 2020-02-13 Wenbo Zhao , Rita Singh

In this paper, a novel architecture for speaker recognition is proposed by cascading speech enhancement and speaker processing. Its aim is to improve speaker recognition performance when speech signals are corrupted by noise. Instead of…

计算与语言 · 计算机科学 2020-05-25 Yanpei Shi , Qiang Huang , Thomas Hain

Teleconferencing is becoming essential during the COVID-19 pandemic. However, in real-world applications, speech quality can deteriorate due to, for example, background interference, noise, or reverberation. To solve this problem, target…

音频与语音处理 · 电气工程与系统科学 2022-05-02 Yicheng Hsu , Yonghan Lee , Mingsian R. Bai

Varying data augmentation policies and regularization over the course of optimization has led to performance improvements over using fixed values. We show that population based training is a useful tool to continuously search those…

计算与语言 · 计算机科学 2020-10-09 Daniel Haziza , Jérémy Rapin , Gabriel Synnaeve

Reducing the interference noise in a monaural noisy speech signal has been a challenging task for many years. Compared to traditional unsupervised speech enhancement methods, e.g., Wiener filtering, supervised approaches, such as algorithms…

声音 · 计算机科学 2017-09-19 Nasser Mohammadiha , Paris Smaragdis , Arne Leijon

This work presents a novel approach for speaker diarization to leverage lexical information provided by automatic speech recognition. We propose a speaker diarization system that can incorporate word-level speaker turn probabilities with…

音频与语音处理 · 电气工程与系统科学 2020-04-16 Tae Jin Park , Kyu J. Han , Jing Huang , Xiaodong He , Bowen Zhou , Panayiotis Georgiou , Shrikanth Narayanan

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

We consider the problem of training speech recognition systems without using any labeled data, under the assumption that the learner can only access to the input utterances and a phoneme language model estimated from a non-overlapping…

音频与语音处理 · 电气工程与系统科学 2018-12-27 Chih-Kuan Yeh , Jianshu Chen , Chengzhu Yu , Dong Yu