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audio Enhancement for Computer Audition—An Iterative Training Paradigm Using Sample Importance

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Journal of Computer Science & Technology 2024年第4期39卷 895-911页

作者： Manuel Milling Shuo Liu Andreas Triantafyllopoulos Ilhan Aslan Björn W.Schuller Chair of Embedded Intelligence for Health Care and Wellbeing University of AugsburgAugsburg 86159Germany Chair of Health Informatics München rechts der IsarTechnical University of MunichMunich 81675Germany Munich Center for Machine Learning Munich 80333Germany Huawei Technologies MunichMunich 80992Germany Munich Data Science Institute Garching 85748Germany Group on Language Audio and MusicImperial College LondonLondon SW72AZU.K.

Neural network models for audio tasks,such as automatic speech recognition(ASR)and acoustic scene classification(ASC),are susceptible to noise contamination for real-life *** improve audio quality,an enhancement module,which can be developed independently,is explicitly used at the front-end of the target audio *** this paper,we present an end-to-end learning solution to jointly optimise the models for audio enhancement(AE)and the subsequent *** guide the optimisation of the AE module towards a target application,and especially to overcome difficult samples,we make use of the sample-wise performance measure as an indication of sample *** experiments,we consider four representative applications to evaluate our training paradigm,i.e.,ASR,speech command recognition(SCR),speech emotion recognition(SER),and *** applications are associated with speech and nonspeech tasks concerning semantic and non-semantic features,transient and global information,and the experimental results indicate that our proposed approach can considerably boost the noise robustness of the models,especially at low signal-to-noise ratios,for a wide range of computer audition tasks in everyday-life noisy environments.

关键词： audio enhancement computer audition joint optimisation multi-task learning voice suppression

丝带振膜与锥形振盆精髓大结合PS audio Apsen FR20

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家庭影院技术 2024年第1期327卷 46-46页

作者：厂家来稿 PS audio

但凡亲身接触过PSaudio新一代旗舰扬声器ApsenFR30的发烧友,对于其洒脱不凡造型、壮阔的音场布局与及高密度乐器结像,必定留下深刻印象,唯独其5英尺身高,加上230英磅体重,并非家中所有成员都能豁然接纳。

关键词：音场振膜 audio 扬声器 PS 高密度

A Novel Robust Zero-Watermarking Algorithm for audio Based on Sparse Representation

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China Communications 2021年第8期18卷 237-248页

作者： Longting Xu Daiyu Huang Xing Guo Wei Rao Yunyun Ji Ruoyi Li Xiaochen Lu College of Information Science and Technology Donghua UniversityShanghai 200000China Tencent Ethereal audio Lab China

Behind the prevalence of multimedia technology,digital copyright disputes are becoming increasingly *** digital watermarking prevention technique against the copyright infringement needs to be improved *** the proposed technologies,zero-watermarking has been favored *** order to improve the robustness of the zero-watermarking,a novel robust audio zerowatermarking method based on sparse representation is *** proposed scheme is mainly based on the K-singular value decomposition(K-SVD)algorithm to construct an optimal over complete dictionary from the background audio *** that,the orthogonal matching pursuit(OMP)algorithm is used to calculate the sparse coefficient of the segmented test audio and generate the corresponding sparse coefficient ***,the mean value of absolute sparse coefficients in the sparse matrix of segmented speech is calculated and selected,and then comparing the mean absolute coefficient of segmented speech with the average value of the selected coefficients to realize the embedding of *** results show that the proposed audio zerowatermarking algorithm based on sparse representation performs effectively in resisting various common *** with the baseline works,the proposed method has better robustness.

关键词： zero-watermarking K-singular value decomposition dictionary learning sparse representtion

audio-Text Multimodal Speech Recognition via Dual-Tower Architecture for Mandarin Air Traffic Control Communications

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Computers, Materials & Continua 2024年第3期78卷 3215-3245页

作者： Shuting Ge Jin Ren Yihua Shi Yujun Zhang Shunzhi Yang Jinfeng Yang School of Computer Science and Software Engineering University of Science and Technology LiaoningAnshan114051China Institute of Applied Artificial Intelligence of the Guangdong-Hong Kong-Macao Greater Bay Area Shenzhen Polytechnic UniversityShenzhen518055China Shenzhen Institutes of Advanced Technology Chinese Academy of SciencesShenzhen518055China Industrial Training centre Shenzhen Polytechnic UniversityShenzhen518055China

In air traffic control communications (ATCC), misunderstandings between pilots and controllers could result in fatal aviation accidents. Fortunately, advanced automatic speech recognition technology has emerged as a promising means of preventing miscommunications and enhancing aviation safety. However, most existing speech recognition methods merely incorporate external language models on the decoder side, leading to insufficient semantic alignment between speech and text modalities during the encoding phase. Furthermore, it is challenging to model acoustic context dependencies over long distances due to the longer speech sequences than text, especially for the extended ATCC data. To address these issues, we propose a speech-text multimodal dual-tower architecture for speech recognition. It employs cross-modal interactions to achieve close semantic alignment during the encoding stage and strengthen its capabilities in modeling auditory long-distance context dependencies. In addition, a two-stage training strategy is elaborately devised to derive semantics-aware acoustic representations effectively. The first stage focuses on pre-training the speech-text multimodal encoding module to enhance inter-modal semantic alignment and aural long-distance context dependencies. The second stage fine-tunes the entire network to bridge the input modality variation gap between the training and inference phases and boost generalization performance. Extensive experiments demonstrate the effectiveness of the proposed speech-text multimodal speech recognition method on the ATCC and AISHELL-1 datasets. It reduces the character error rate to 6.54% and 8.73%, respectively, and exhibits substantial performance gains of 28.76% and 23.82% compared with the best baseline model. The case studies indicate that the obtained semantics-aware acoustic representations aid in accurately recognizing terms with similar pronunciations but distinctive semantics. The research provides a novel modeling pa

关键词： Speech-text multimodal automatic speech recognition semantic alignment air traffic control communications dual-tower architecture

Transparent and Ultra-lightweight Design for Ultra-Broadband Asymmetric Transmission of Airborne Sound

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Chinese Physics Letters 2018年第2期35卷 34-37页

作者： Jie Hu Bin Liang Xiao-Jun Qiu Key Laboratory of Modern Acoustics(MOE) Institute of AcousticsNanjing UniversityNanjing 210023 College of Information Science and Technology Nanjing Forestry UniversityNanjing 210000 centre for audio Acoustics and VibrationFaculty of Engineering and ITUniversity of Technology SydneyAustrlia

Acoustic one-way manipulations have recently attracted significant attention due to the deep implications in many diverse fields such as biomedical imaging and ***,the previous mechanisms of asymmetric manipulation of airborne sound need to use elaborate heavyweight structures and only work in certain frequency *** propose a mechanism for designing an ultra-lightweight and optically transparent structure with asymmetric transmission property for normally incident plane *** of fabricating solids into complicated artificial structures with limited bandwidth and heavy asymmetric shape which allows the incident plane wave weight,we simply use xenon to fill a spatial region of to pass along one direction while reflecting the reversed wave regardless of *** demonstrate both analytically and numerically its effectiveness of producing highly-asymmetric transmission within an ultra-broad *** design offers new possibility for the design of one-way devices and may have far-reaching impact on various scenarios such as noise control.

关键词： Acoustic fields

Adaptive progressive download based on the MPEG-4 file format

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Journal of Zhejiang University-Science A(Applied Physics & Engineering) 2006年第Z1期7卷 106-111页

作者： FRBER Nikolaus DHLA Stefan ISSING Jochen Multimedia Transport Dept.audio Fraunhofer Institute for Integrated Circuits IISErlangen 91058Germany

In this paper we describe how progressive download and adaptive streaming can be combined into a simple and efficient streaming framework. Based on the MPEG-4 file format (MP4) we use HTTP for transport and argue that these two components are sufficient for specifying an open streaming architecture. The client selects appropriate chunks from the MP4 file to be transferred based on (1) the header information (i.e. the 'moov' box) in the first part of the file and (2) his observation of network throughput. The framework is completely client driven which allows for better server scalability and reduces signalling overhead. We discuss architecture and implementation issues such as complexity, interoperability and scalability and compare to 3GPP PSS Re1-6 adaptive streaming when appropriate. Measurements from a simple MP4/HTTP streaming client are presented showing that appropriate chunks are selected such that increased reliability is achieved.

关键词： HTTP-streaming Adaptive progressive download MPEG-4 file format (MP4) Moov box

Weighting Function Modification Used for Phase Transform-Based Time Delay Estimation

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China Communications 2022年第11期19卷 241-256页

作者： Xue Yang Changchun Bao Zihao Cui Speech and audio Signal Processing Laboratory Faculty of Information TechnologyBeijing University of TechnologyBeijing 100124China

Generalized cross-correlation is considered as the most straightforward time delay estimation *** on various weighting function,different methods were derived and a straightforward method,named phase transform(PHAT)has been widely *** is well-known for its robustness to reverberation and its sensitivity to noise,which is partly due to the fact that PHAT distributes same weights to the frequencies dominated by signal or *** alleviate this problem,two weighting functions are proposed in this *** taking a posteriori signal-to-noise ratio(SNR)into account to classify reliable and unreliable frequencies,different weights could be *** first proposed weighting function borrows the idea of binary mask and distributes same weights to frequencies in same set,whereas,the second one assigns weights based on coherence *** showed the robustness of proposed methods to reverberation and noise for improving the performance of time delay estimation through various criteria.

关键词： time delay estimation generalized crosscorrelation PHAT a posteriori SNR coherence function

DRA audio Coding Standard

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Chinese Journal of Electronics 2014年第3期23卷 521-526页

作者： MA Wenhua XU Jing MA Yuanzhe YOU Yuli Department of Computers Cisco School of InformaticsGuangdong University of Foreign Studies Department of Science and Technology Guangdong Rising Assets Management Co. Department of Biomedical Science and Technology South China University of Technology Guangdong Provincial Key Laboratary for Digital audio Technology

DRA(Dynamic resolution adaptation)audio coding standard was shown to deploy transientlocalized MDCT to effectively suppress pre-echo artifacts and statistic allocation of codebooks to improve the compression efficiency of Huffman coding. Its quantizers and Huffman codebooks are designed in such a way that a signal path of 24 bits is provided throughout the codec so that high audio quality can be delivered if bit rate *** simple, it delivers state-of-the-arts compression efficiency as shown by five rounds of ITU-R BS.11116 compliant subjective listening tests.

关键词： audio coding Standard Listening test Modified discrete cosine transform(MDCT) Huffman coding

Delay-Based Cross-Layer QoS Scheme for Video Streaming in Wireless Ad Hoc Networks

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China Communications 2018年第9期15卷 215-234页

作者： Shu Fan Honglin Zhao Communication Research Center Harbin Institute of Technology audio-Visual and Image Technology Department Criminal Investigation Police University of China

With correlating with human perception, quality of experience(Qo E) is also an important measurement in evaluation of video quality in addition to quality of service(Qo S). A cross-layer scheme based on Lyapunov optimization framework for H.264/AVC video streaming over wireless Ad hoc networks is proposed, with increasing both Qo E and Qo S performances. Different from existing works, this scheme routes and schedules video packets according to the statuses of the frame buffers at the destination nodes to reduce buffer underflows and to increase video playout continuity. The waiting time of head-ofline packets of data queues are considered in routing and scheduling to reduce the average end-to-end delay of video sessions. Different types of packets are allocated with different priorities according to their generated rates under H.264/AVC. To reduce the computational complexity, a distributed media access control policy and a power control algorithm cooperating with the media access policy are proposed. Simulation results show that, compared with existing schemes, this scheme can improve both the Qo S and Qo E performances. The average peak signal-to-noise ratio(PSNR) of the received video streams is also increased.

关键词： cross-layer control Lyapunov optimization wireless ad hoc networks video streaming