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ORCID23rd Interspeech 2022: Incheon, Korea
- Hanseok Ko, John H. L. Hansen:
23rd Annual Conference of the International Speech Communication Association, Interspeech 2022, Incheon, Korea, September 18-22, 2022. ISCA 2022
Speech Synthesis: Toward end-to-end synthesis
- Hyunjae Cho, Wonbin Jung, Junhyeok Lee
, Sang Hoon Woo:
SANE-TTS: Stable And Natural End-to-End Multilingual Text-to-Speech. 1-5 - Hanbin Bae, Young-Sun Joo:
Enhancement of Pitch Controllability using Timbre-Preserving Pitch Augmentation in FastPitch. 6-10 - Martin Lenglet, Olivier Perrotin, Gérard Bailly:
Speaking Rate Control of end-to-end TTS Models by Direct Manipulation of the Encoder's Output Embeddings. 11-15 - Yooncheol Ju, Ilhwan Kim, Hongsun Yang, Ji-Hoon Kim, Byeongyeol Kim, Soumi Maiti, Shinji Watanabe:
TriniTTS: Pitch-controllable End-to-end TTS without External Aligner. 16-20 - Dan Lim, Sunghee Jung, Eesung Kim:
JETS: Jointly Training FastSpeech2 and HiFi-GAN for End to End Text to Speech. 21-25
Technology for Disordered Speech
- Rosanna Turrisi, Leonardo Badino:
Interpretable dysarthric speaker adaptation based on optimal-transport. 26-30 - Zhengjun Yue, Erfan Loweimi, Heidi Christensen, Jon Barker, Zoran Cvetkovic:
Dysarthric Speech Recognition From Raw Waveform with Parametric CNNs. 31-35 - Luke Prananta, Bence Mark Halpern, Siyuan Feng, Odette Scharenborg:
The Effectiveness of Time Stretching for Enhancing Dysarthric Speech for Improved Dysarthric Speech Recognition. 36-40 - Lester Phillip Violeta, Wen-Chin Huang, Tomoki Toda:
Investigating Self-supervised Pretraining Frameworks for Pathological Speech Recognition. 41-45 - Chitralekha Bhat, Ashish Panda, Helmer Strik:
Improved ASR Performance for Dysarthric Speech Using Two-stage DataAugmentation. 46-50 - Abner Hernandez, Paula Andrea Pérez-Toro, Elmar Nöth, Juan Rafael Orozco-Arroyave, Andreas K. Maier, Seung Hee Yang:
Cross-lingual Self-Supervised Speech Representations for Improved Dysarthric Speech Recognition. 51-55
Neural Network Training Methods for ASR I
- Mun-Hak Lee, Joon-Hyuk Chang, Sang-Eon Lee, Ju-Seok Seong, Chanhee Park, Haeyoung Kwon:
Regularizing Transformer-based Acoustic Models by Penalizing Attention Weights. 56-60 - David M. Chan, Shalini Ghosh:
Content-Context Factorized Representations for Automated Speech Recognition. 61-65 - Georgios Karakasidis, Tamás Grósz, Mikko Kurimo:
Comparison and Analysis of New Curriculum Criteria for End-to-End ASR. 66-70 - Deepak Baby, Pasquale D'Alterio, Valentin Mendelev:
Incremental learning for RNN-Transducer based speech recognition models. 71-75 - Andrew Hard, Kurt Partridge, Neng Chen, Sean Augenstein, Aishanee Shah, Hyun Jin Park, Alex Park, Sara Ng, Jessica Nguyen, Ignacio López-Moreno, Rajiv Mathews, Françoise Beaufays:
Production federated keyword spotting via distillation, filtering, and joint federated-centralized training. 76-80
Acoustic Phonetics and Prosody
- Jieun Song, Hae-Sung Jeon, Jieun Kiaer:
Use of prosodic and lexical cues for disambiguating wh-words in Korean. 81-85 - Vinicius Ribeiro, Yves Laprie:
Autoencoder-Based Tongue Shape Estimation During Continuous Speech. 86-90 - Giuseppe Magistro, Claudia Crocco:
Phonetic erosion and information structure in function words: the case of mia. 91-95 - Miran Oh, Yoon-Jeong Lee:
Dynamic Vertical Larynx Actions Under Prosodic Focus. 96-100 - Leah Bradshaw, Eleanor Chodroff, Lena A. Jäger, Volker Dellwo:
Fundamental Frequency Variability over Time in Telephone Interactions. 101-105
Spoken Machine Translation
- Ioannis Tsiamas, Gerard I. Gállego, José A. R. Fonollosa, Marta R. Costa-jussà:
SHAS: Approaching optimal Segmentation for End-to-End Speech Translation. 106-110 - Jinming Zhao, Hao Yang, Gholamreza Haffari, Ehsan Shareghi:
M-Adapter: Modality Adaptation for End-to-End Speech-to-Text Translation. 111-115 - Mohd Abbas Zaidi, Beomseok Lee, Sangha Kim, Chanwoo Kim:
Cross-Modal Decision Regularization for Simultaneous Speech Translation. 116-120 - Ryo Fukuda, Katsuhito Sudoh, Satoshi Nakamura:
Speech Segmentation Optimization using Segmented Bilingual Speech Corpus for End-to-end Speech Translation. 121-125 - Kirandevraj R, Vinod Kumar Kurmi, Vinay P. Namboodiri, C. V. Jawahar:
Generalized Keyword Spotting using ASR embeddings. 126-130
(Multimodal) Speech Emotion Recognition I
- Youngdo Ahn, Sung Joo Lee, Jong Won Shin:
Multi-Corpus Speech Emotion Recognition for Unseen Corpus Using Corpus-Wise Weights in Classification Loss. 131-135 - Junghun Kim, Yoojin An, Jihie Kim:
Improving Speech Emotion Recognition Through Focus and Calibration Attention Mechanisms. 136-140 - Joosung Lee:
The Emotion is Not One-hot Encoding: Learning with Grayscale Label for Emotion Recognition in Conversation. 141-145 - Andreas Triantafyllopoulos, Johannes Wagner, Hagen Wierstorf, Maximilian Schmitt, Uwe D. Reichel, Florian Eyben, Felix Burkhardt, Björn W. Schuller:
Probing speech emotion recognition transformers for linguistic knowledge. 146-150 - Navin Raj Prabhu, Guillaume Carbajal, Nale Lehmann-Willenbrock, Timo Gerkmann:
End-To-End Label Uncertainty Modeling for Speech-based Arousal Recognition Using Bayesian Neural Networks. 151-155 - Matthew Perez, Mimansa Jaiswal, Minxue Niu, Cristina Gorrostieta, Matthew Roddy, Kye Taylor, Reza Lotfian, John Kane, Emily Mower Provost:
Mind the gap: On the value of silence representations to lexical-based speech emotion recognition. 156-160 - Huang-Cheng Chou, Chi-Chun Lee, Carlos Busso:
Exploiting Co-occurrence Frequency of Emotions in Perceptual Evaluations To Train A Speech Emotion Classifier. 161-165 - Hira Dhamyal, Bhiksha Raj, Rita Singh:
Positional Encoding for Capturing Modality Specific Cadence for Emotion Detection. 166-170
Dereverberation, Noise Reduction, and Speaker Extraction
- Tuan Vu Ho, Maori Kobayashi, Masato Akagi:
Speak Like a Professional: Increasing Speech Intelligibility by Mimicking Professional Announcer Voice with Voice Conversion. 171-175 - Tuan Vu Ho, Quoc Huy Nguyen, Masato Akagi, Masashi Unoki:
Vector-quantized Variational Autoencoder for Phase-aware Speech Enhancement. 176-180 - Minseung Kim, Hyungchan Song, Sein Cheong, Jong Won Shin:
iDeepMMSE: An improved deep learning approach to MMSE speech and noise power spectrum estimation for speech enhancement. 181-185 - Kuo-Hsuan Hung, Szu-Wei Fu, Huan-Hsin Tseng, Hsin-Tien Chiang, Yu Tsao, Chii-Wann Lin:
Boosting Self-Supervised Embeddings for Speech Enhancement. 186-190 - Seorim Hwang, Youngcheol Park, Sungwook Park:
Monoaural Speech Enhancement Using a Nested U-Net with Two-Level Skip Connections. 191-195 - Hannah Muckenhirn, Aleksandr Safin, Hakan Erdogan, Felix de Chaumont Quitry, Marco Tagliasacchi, Scott Wisdom, John R. Hershey:
CycleGAN-based Unpaired Speech Dereverberation. 196-200 - Ashutosh Pandey, DeLiang Wang:
Attentive Training: A New Training Framework for Talker-independent Speaker Extraction. 201-205 - Tyler Vuong, Richard M. Stern:
Improved Modulation-Domain Loss for Neural-Network-based Speech Enhancement. 206-210 - Chiang-Jen Peng, Yun-Ju Chan, Yih-Liang Shen, Cheng Yu, Yu Tsao, Tai-Shih Chi:
Perceptual Characteristics Based Multi-objective Model for Speech Enhancement. 211-215 - Marc Delcroix, Keisuke Kinoshita, Tsubasa Ochiai, Katerina Zmolíková, Hiroshi Sato, Tomohiro Nakatani:
Listen only to me! How well can target speech extraction handle false alarms? 216-220 - Hao Shi, Longbiao Wang, Sheng Li, Jianwu Dang, Tatsuya Kawahara:
Monaural Speech Enhancement Based on Spectrogram Decomposition for Convolutional Neural Network-sensitive Feature Extraction. 221-225 - Jean-Marie Lemercier, Joachim Thiemann, Raphael Koning, Timo Gerkmann:
Neural Network-augmented Kalman Filtering for Robust Online Speech Dereverberation in Noisy Reverberant Environments. 226-230
Source Separation II
- Nicolás Schmidt, Jordi Pons, Marius Miron:
PodcastMix: A dataset for separating music and speech in podcasts. 231-235 - Kohei Saijo, Robin Scheibler:
Independence-based Joint Dereverberation and Separation with Neural Source Model. 236-240 - Kohei Saijo, Robin Scheibler:
Spatial Loss for Unsupervised Multi-channel Source Separation. 241-245 - Samuel Bellows, Timothy W. Leishman:
Effect of Head Orientation on Speech Directivity. 246-250 - Kohei Saijo, Tetsuji Ogawa:
Unsupervised Training of Sequential Neural Beamformer Using Coarsely-separated and Non-separated Signals. 251-255 - Marvin Borsdorf, Kevin Scheck, Haizhou Li, Tanja Schultz:
Blind Language Separation: Disentangling Multilingual Cocktail Party Voices by Language. 256-260 - Mateusz Guzik, Konrad Kowalczyk:
NTF of Spectral and Spatial Features for Tracking and Separation of Moving Sound Sources in Spherical Harmonic Domain. 261-265 - Jack Deadman, Jon Barker:
Modelling Turn-taking in Multispeaker Parties for Realistic Data Simulation. 266-270 - Christoph Böddeker, Tobias Cord-Landwehr, Thilo von Neumann, Reinhold Haeb-Umbach:
An Initialization Scheme for Meeting Separation with Spatial Mixture Models. 271-275 - Seongkyu Mun, Dhananjaya Gowda, Jihwan Lee, Changwoo Han, Dokyun Lee, Chanwoo Kim:
Prototypical speaker-interference loss for target voice separation using non-parallel audio samples. 276-280
Embedding and Network Architecture for Speaker Recognition
- Pierre-Michel Bousquet, Mickael Rouvier, Jean-François Bonastre:
Reliability criterion based on learning-phase entropy for speaker recognition with neural network. 281-285 - Bei Liu, Zhengyang Chen, Yanmin Qian:
Attentive Feature Fusion for Robust Speaker Verification. 286-290 - Bei Liu, Zhengyang Chen, Yanmin Qian:
Dual Path Embedding Learning for Speaker Verification with Triplet Attention. 291-295 - Bei Liu, Zhengyang Chen, Shuai Wang, Haoyu Wang, Bing Han, Yanmin Qian:
DF-ResNet: Boosting Speaker Verification Performance with Depth-First Design. 296-300 - Ruida Li, Shuo Fang, Chenguang Ma, Liang Li:
Adaptive Rectangle Loss for Speaker Verification. 301-305 - Yang Zhang, Zhiqiang Lv, Haibin Wu, Shanshan Zhang, Pengfei Hu, Zhiyong Wu, Hung-yi Lee, Helen Meng:
MFA-Conformer: Multi-scale Feature Aggregation Conformer for Automatic Speaker Verification. 306-310 - Leying Zhang, Zhengyang Chen, Yanmin Qian:
Enroll-Aware Attentive Statistics Pooling for Target Speaker Verification. 311-315 - Yusheng Tian, Jingyu Li, Tan Lee:
Transport-Oriented Feature Aggregation for Speaker Embedding Learning. 316-320 - Mufan Sang, John H. L. Hansen:
Multi-Frequency Information Enhanced Channel Attention Module for Speaker Representation Learning. 321-325 - Linjun Cai, Yuhong Yang, Xufeng Chen, Weiping Tu, Hongyang Chen:
CS-CTCSCONV1D: Small footprint speaker verification with channel split time-channel-time separable 1-dimensional convolution. 326-330 - Pengqi Li, Lantian Li, Askar Hamdulla, Dong Wang:
Reliable Visualization for Deep Speaker Recognition. 331-335 - Zhiyuan Peng, Xuanji He, Ke Ding, Tan Lee, Guanglu Wan:
Unifying Cosine and PLDA Back-ends for Speaker Verification. 336-340 - Yuheng Wei, Junzhao Du, Hui Liu, Qian Wang:
CTFALite: Lightweight Channel-specific Temporal and Frequency Attention Mechanism for Enhancing the Speaker Embedding Extractor. 341-345
Speech Representation II
- Weidong Chen, Xiaofen Xing, Xiangmin Xu, Jianxin Pang, Lan Du:
SpeechFormer: A Hierarchical Efficient Framework Incorporating the Characteristics of Speech. 346-350 - David Feinberg:
VoiceLab: Software for Fully Reproducible Automated Voice Analysis. 351-355 - Joel Shor, Subhashini Venugopalan:
TRILLsson: Distilled Universal Paralinguistic Speech Representations. 356-360 - Nan Li, Meng Ge, Longbiao Wang, Masashi Unoki, Sheng Li, Jianwu Dang:
Global Signal-to-noise Ratio Estimation Based on Multi-subband Processing Using Convolutional Neural Network. 361-365 - Mostafa Sadeghi, Paul Magron:
A Sparsity-promoting Dictionary Model for Variational Autoencoders. 366-370 - Yan Zhao, Jincen Wang, Ru Ye, Yuan Zong, Wenming Zheng, Li Zhao:
Deep Transductive Transfer Regression Network for Cross-Corpus Speech Emotion Recognition. 371-375 - John H. L. Hansen, Zhenyu Wang:
Audio Anti-spoofing Using Simple Attention Module and Joint Optimization Based on Additive Angular Margin Loss and Meta-learning. 376-380 - Boris Bergsma, Minhao Yang, Milos Cernak:
PEAF: Learnable Power Efficient Analog Acoustic Features for Audio Recognition. 381-385 - Gasser Elbanna, Alice Biryukov, Neil Scheidwasser-Clow, Lara Orlandic, Pablo Mainar, Mikolaj Kegler, Pierre Beckmann, Milos Cernak:
Hybrid Handcrafted and Learnable Audio Representation for Analysis of Speech Under Cognitive and Physical Load. 386-390 - Shijun Wang, Hamed Hemati, Jón Guðnason, Damian Borth:
Generative Data Augmentation Guided by Triplet Loss for Speech Emotion Recognition. 391-395 - Sarthak Yadav, Neil Zeghidour:
Learning neural audio features without supervision. 396-400 - Yixuan Zhang, Heming Wang, DeLiang Wang:
Densely-connected Convolutional Recurrent Network for Fundamental Frequency Estimation in Noisy Speech. 401-405 - Abu Zaher Md Faridee, Hannes Gamper:
Predicting label distribution improves non-intrusive speech quality estimation. 406-410 - Takanori Ashihara, Takafumi Moriya, Kohei Matsuura, Tomohiro Tanaka:
Deep versus Wide: An Analysis of Student Architectures for Task-Agnostic Knowledge Distillation of Self-Supervised Speech Models. 411-415 - Abdul Hameed Azeemi, Ihsan Ayyub Qazi, Agha Ali Raza:
Dataset Pruning for Resource-constrained Spoofed Audio Detection. 416-420
Speech Synthesis: Linguistic Processing, Paradigms and Other Topics II
- Jaesung Tae, Hyeongju Kim, Taesu Kim:
EdiTTS: Score-based Editing for Controllable Text-to-Speech. 421-425 - Jie Chen, Changhe Song, Deyi Tuo, Xixin Wu, Shiyin Kang, Zhiyong Wu, Helen Meng:
Improving Mandarin Prosodic Structure Prediction with Multi-level Contextual Information. 426-430 - Zalan Borsos, Matthew Sharifi, Marco Tagliasacchi:
SpeechPainter: Text-conditioned Speech Inpainting. 431-435 - Song Zhang, Ken Zheng, Xiaoxu Zhu, Baoxiang Li:
A polyphone BERT for Polyphone Disambiguation in Mandarin Chinese. 436-440 - Mutian He, Jingzhou Yang, Lei He, Frank K. Soong:
Neural Lexicon Reader: Reduce Pronunciation Errors in End-to-end TTS by Leveraging External Textual Knowledge. 441-445 - Jian Zhu, Cong Zhang, David Jurgens:
ByT5 model for massively multilingual grapheme-to-phoneme conversion. 446-450 - Puneet Mathur, Franck Dernoncourt, Quan Hung Tran, Jiuxiang Gu, Ani Nenkova, Vlad I. Morariu, Rajiv Jain, Dinesh Manocha:
DocLayoutTTS: Dataset and Baselines for Layout-informed Document-level Neural Speech Synthesis. 451-455 - Guangyan Zhang, Kaitao Song, Xu Tan, Daxin Tan, Yuzi Yan, Yanqing Liu, Gang Wang, Wei Zhou, Tao Qin, Tan Lee, Sheng Zhao:
Mixed-Phoneme BERT: Improving BERT with Mixed Phoneme and Sup-Phoneme Representations for Text to Speech. 456-460 - Junrui Ni, Liming Wang, Heting Gao, Kaizhi Qian, Yang Zhang, Shiyu Chang, Mark Hasegawa-Johnson:
Unsupervised Text-to-Speech Synthesis by Unsupervised Automatic Speech Recognition. 461-465 - Tho Nguyen Duc Tran, The Chuong Chu, Vu Hoang, Trung Huu Bui, Steven Hung Quoc Truong:
An Efficient and High Fidelity Vietnamese Streaming End-to-End Speech Synthesis. 466-470 - Cassia Valentini-Botinhao, Manuel Sam Ribeiro, Oliver Watts, Korin Richmond, Gustav Eje Henter:
Predicting pairwise preferences between TTS audio stimuli using parallel ratings data and anti-symmetric twin neural networks. 471-475 - Zikai Chen, Lin Wu, Junjie Pan, Xiang Yin:
An Automatic Soundtracking System for Text-to-Speech Audiobooks. 476-480 - Daxin Tan, Guangyan Zhang, Tan Lee:
Environment Aware Text-to-Speech Synthesis. 481-485 - Artem Ploujnikov, Mirco Ravanelli:
SoundChoice: Grapheme-to-Phoneme Models with Semantic Disambiguation. 486-490 - Evelina Bakhturina, Yang Zhang, Boris Ginsburg:
Shallow Fusion of Weighted Finite-State Transducer and Language Model for Text Normalization. 491-495 - Yogesh Virkar, Marcello Federico, Robert Enyedi, Roberto Barra-Chicote:
Prosodic alignment for off-screen automatic dubbing. 496-500 - Qibing Bai, Tom Ko, Yu Zhang:
A Study of Modeling Rising Intonation in Cantonese Neural Speech Synthesis. 501-505 - Hirokazu Kameoka, Takuhiro Kaneko, Shogo Seki, Kou Tanaka:
CAUSE: Crossmodal Action Unit Sequence Estimation from Speech. 506-510 - Binu Nisal Abeysinghe, Jesin James, Catherine I. Watson, Felix Marattukalam:
Visualising Model Training via Vowel Space for Text-To-Speech Systems. 511-515
Other Topics in Speech Recognition
- Aaqib Saeed:
Binary Early-Exit Network for Adaptive Inference on Low-Resource Devices. 516-520 - Naoyuki Kanda, Jian Wu, Yu Wu, Xiong Xiao, Zhong Meng, Xiaofei Wang, Yashesh Gaur, Zhuo Chen, Jinyu Li, Takuya Yoshioka:
Streaming Speaker-Attributed ASR with Token-Level Speaker Embeddings. 521-525 - Naoki Makishima, Satoshi Suzuki, Atsushi Ando, Ryo Masumura:
Speaker consistency loss and step-wise optimization for semi-supervised joint training of TTS and ASR using unpaired text data. 526-530 - Yi-Kai Zhang, Da-Wei Zhou, Han-Jia Ye, De-Chuan Zhan:
Audio-Visual Generalized Few-Shot Learning with Prototype-Based Co-Adaptation. 531-535 - Junteng Jia, Jay Mahadeokar, Weiyi Zheng, Yuan Shangguan, Ozlem Kalinli, Frank Seide:
Federated Domain Adaptation for ASR with Full Self-Supervision. 536-540 - Longfei Yang, Wenqing Wei, Sheng Li, Jiyi Li, Takahiro Shinozaki:
Augmented Adversarial Self-Supervised Learning for Early-Stage Alzheimer's Speech Detection. 541-545 - Zvi Kons, Hagai Aronowitz, Edmilson da Silva Morais, Matheus Damasceno, Hong-Kwang Kuo, Samuel Thomas, George Saon:
Extending RNN-T-based speech recognition systems with emotion and language classification. 546-549 - Alexandra Antonova, Evelina Bakhturina, Boris Ginsburg:
Thutmose Tagger: Single-pass neural model for Inverse Text Normalization. 550-554 - Yeonjin Cho, Sara Ng, Trang Tran, Mari Ostendorf:
Leveraging Prosody for Punctuation Prediction of Spontaneous Speech. 555-559 - Fan Yu, Zhihao Du, Shiliang Zhang, Yuxiao Lin, Lei Xie:
A Comparative Study on Speaker-attributed Automatic Speech Recognition in Multi-party Meetings. 560-564
Audio Deep PLC (Packet Loss Concealment) Challenge
- Yuansheng Guan, Guochen Yu, Andong Li, Chengshi Zheng, Jie Wang:
TMGAN-PLC: Audio Packet Loss Concealment using Temporal Memory Generative Adversarial Network. 565-569 - Jean-Marc Valin, Ahmed Mustafa, Christopher Montgomery, Timothy B. Terriberry, Michael Klingbeil, Paris Smaragdis, Arvindh Krishnaswamy:
Real-Time Packet Loss Concealment With Mixed Generative and Predictive Model. 570-574 - Baiyun Liu, Qi Song, Mingxue Yang, Wuwen Yuan, Tianbao Wang:
PLCNet: Real-time Packet Loss Concealment with Semi-supervised Generative Adversarial Network. 575-579 - Lorenz Diener, Sten Sootla, Solomiya Branets, Ando Saabas, Robert Aichner, Ross Cutler:
INTERSPEECH 2022 Audio Deep Packet Loss Concealment Challenge. 580-584 - Nan Li, Xiguang Zheng, Chen Zhang, Liang Guo, Bing Yu:
End-to-End Multi-Loss Training for Low Delay Packet Loss Concealment. 585-589
Robust Speaker Recognition
- Ju-ho Kim, Jungwoo Heo, Hye-jin Shim, Ha-Jin Yu:
Extended U-Net for Speaker Verification in Noisy Environments. 590-594 - Seunghan Yang, Debasmit Das, Janghoon Cho, Hyoungwoo Park, Sungrack Yun:
Domain Agnostic Few-shot Learning for Speaker Verification. 595-599 - Qiongqiong Wang, Kong Aik Lee, Tianchi Liu:
Scoring of Large-Margin Embeddings for Speaker Verification: Cosine or PLDA? 600-604 - Themos Stafylakis, Ladislav Mosner, Oldrich Plchot, Johan Rohdin, Anna Silnova, Lukás Burget, Jan Cernocký:
Training speaker embedding extractors using multi-speaker audio with unknown speaker boundaries. 605-609 - Chau Luu, Steve Renals, Peter Bell:
Investigating the contribution of speaker attributes to speaker separability using disentangled speaker representations. 610-614 - Saurabh Kataria, Jesús Villalba, Laureano Moro-Velázquez, Najim Dehak:
Joint domain adaptation and speech bandwidth extension using time-domain GANs for speaker verification. 615-619
Speech Production
- Tsukasa Yoshinaga, Kikuo Maekawa, Akiyoshi Iida:
Variability in Production of Non-Sibilant Fricative [ç] in /hi/. 620-624 - Sathvik Udupa, Aravind Illa, Prasanta Kumar Ghosh:
Streaming model for Acoustic to Articulatory Inversion with transformer networks. 625-629 - Tsiky Rakotomalala, Pierre Baraduc, Pascal Perrier:
Trajectories predicted by optimal speech motor control using LSTM networks. 630-634 - Daniel R. van Niekerk, Anqi Xu, Branislav Gerazov, Paul Konstantin Krug, Peter Birkholz, Yi Xu:
Exploration strategies for articulatory synthesis of complex syllable onsets. 635-639 - Yoonjeong Lee, Jody Kreiman:
Linguistic versus biological factors governing acoustic voice variation. 640-643 - Takayuki Nagamine:
Acquisition of allophonic variation in second language speech: An acoustic and articulatory study of English laterals by Japanese speakers. 644-648
Speech Quality Assessment
- Pranay Manocha, Anurag Kumar, Buye Xu, Anjali Menon, Israel Dejene Gebru, Vamsi Krishna Ithapu, Paul Calamia:
SAQAM: Spatial Audio Quality Assessment Metric. 649-653 - Pranay Manocha, Anurag Kumar:
Speech Quality Assessment through MOS using Non-Matching References. 654-658 - Hideki Kawahara, Kohei Yatabe, Ken-Ichi Sakakibara, Tatsuya Kitamura, Hideki Banno, Masanori Morise:
An objective test tool for pitch extractors' response attributes. 659-663 - Kai Li, Sheng Li, Xugang Lu, Masato Akagi, Meng Liu, Lin Zhang, Chang Zeng, Longbiao Wang, Jianwu Dang, Masashi Unoki:
Data Augmentation Using McAdams-Coefficient-Based Speaker Anonymization for Fake Audio Detection. 664-668 - Salah Zaiem, Titouan Parcollet, Slim Essid:
Automatic Data Augmentation Selection and Parametrization in Contrastive Self-Supervised Speech Representation Learning. 669-673 - Deebha Mumtaz, Ajit Jena, Vinit Jakhetiya, Karan Nathwani, Sharath Chandra Guntuku:
Transformer-based quality assessment model for generalized user-generated multimedia audio content. 674-678
Language Modeling and Lexical Modeling for ASR
- Christophe Van Gysel, Mirko Hannemann, Ernest Pusateri, Youssef Oualil, Ilya Oparin:
Space-Efficient Representation of Entity-centric Query Language Models. 679-683 - Saket Dingliwal, Ashish Shenoy, Sravan Bodapati, Ankur Gandhe, Ravi Teja Gadde, Katrin Kirchhoff:
Domain Prompts: Towards memory and compute efficient domain adaptation of ASR systems. 684-688 - W. Ronny Huang, Cal Peyser, Tara N. Sainath, Ruoming Pang, Trevor D. Strohman, Shankar Kumar:
Sentence-Select: Large-Scale Language Model Data Selection for Rare-Word Speech Recognition. 689-693 - Theresa Breiner, Swaroop Ramaswamy, Ehsan Variani, Shefali Garg, Rajiv Mathews, Khe Chai Sim, Kilol Gupta, Mingqing Chen, Lara McConnaughey:
UserLibri: A Dataset for ASR Personalization Using Only Text. 694-698 - Chin-Yueh Chien, Kuan-Yu Chen:
A BERT-based Language Modeling Framework. 699-703
Challenges and Opportunities for Signal Processing and Machine Learning for Multiple Smart Devices
- Yoshiki Masuyama, Kouei Yamaoka, Nobutaka Ono:
Joint Optimization of Sampling Rate Offsets Based on Entire Signal Relationship Among Distributed Microphones. 704-708 - Gregory Ciccarelli, Jarred Barber, Arun Nair, Israel Cohen, Tao Zhang:
Challenges and Opportunities in Multi-device Speech Processing. 709-713 - Ameya Agaskar:
Practical Over-the-air Perceptual AcousticWatermarking. 714-718 - Timm Koppelmann, Luca Becker, Alexandru Nelus, Rene Glitza, Lea Schönherr, Rainer Martin:
Clustering-based Wake Word Detection in Privacy-aware Acoustic Sensor Networks. 719-723 - Francesco Nespoli, Daniel Barreda, Patrick A. Naylor:
Relative Acoustic Features for Distance Estimation in Smart-Homes. 724-728 - Ashutosh Pandey, Buye Xu, Anurag Kumar, Jacob Donley, Paul Calamia, DeLiang Wang:
Time-domain Ad-hoc Array Speech Enhancement Using a Triple-path Network. 729-733
Speech Processing & Measurement
- Arne-Lukas Fietkau, Simon Stone, Peter Birkholz:
Relationship between the acoustic time intervals and tongue movements of German diphthongs. 734-738 - Sanae Matsui, Kyoji Iwamoto, Reiko Mazuka:
Development of allophonic realization until adolescence: A production study of the affricate-fricative variation of /z/ among Japanese children. 739-743 - Chung Soo Ahn, L. L. Chamara Kasun, Sunil Sivadas, Jagath C. Rajapakse:
Recurrent multi-head attention fusion network for combining audio and text for speech emotion recognition. 744-748 - Louise Coppieters de Gibson, Philip N. Garner:
Low-Level Physiological Implications of End-to-End Learning for Speech Recognition. 749-753 - Carolina Lins Machado, Volker Dellwo, Lei He:
Idiosyncratic lingual articulation of American English /æ/ and /ɑ/ using network analysis. 754-758 - Teruki Toya, Wenyu Zhu, Maori Kobayashi, Kenichi Nakamura, Masashi Unoki:
Method for improving the word intelligibility of presented speech using bone-conduction headphones. 759-763 - Debasish Ray Mohapatra, Mario Fleischer, Victor Zappi, Peter Birkholz, Sidney S. Fels:
Three-dimensional finite-difference time-domain acoustic analysis of simplified vocal tract shapes. 764-768 - Dorina De Jong, Aldo Pastore, Noël Nguyen, Alessandro D'Ausilio:
Speech imitation skills predict automatic phonetic convergence: a GMM-UBM study on L2. 769-773 - Marc-Antoine Georges, Jean-Luc Schwartz, Thomas Hueber:
Self-supervised speech unit discovery from articulatory and acoustic features using VQ-VAE. 774-778 - Peter Wu, Shinji Watanabe, Louis Goldstein, Alan W. Black, Gopala Krishna Anumanchipalli:
Deep Speech Synthesis from Articulatory Representations. 779-783 - Monica Ashokumar, Jean-Luc Schwartz, Takayuki Ito:
Orofacial somatosensory inputs in speech perceptual training modulate speech production. 784-787
Speech Synthesis: Acoustic Modeling and Neural Waveform Generation I
- Minchan Kim, Myeonghun Jeong, Byoung Jin Choi, Sunghwan Ahn, Joun Yeop Lee, Nam Soo Kim:
Transfer Learning Framework for Low-Resource Text-to-Speech using a Large-Scale Unlabeled Speech Corpus. 788-792 - Takaaki Saeki, Kentaro Tachibana, Ryuichi Yamamoto:
DRSpeech: Degradation-Robust Text-to-Speech Synthesis with Frame-Level and Utterance-Level Acoustic Representation Learning. 793-797 - Kentaro Mitsui, Kei Sawada:
MSR-NV: Neural Vocoder Using Multiple Sampling Rates. 798-802 - Yuma Koizumi, Heiga Zen, Kohei Yatabe, Nanxin Chen, Michiel Bacchiani:
SpecGrad: Diffusion Probabilistic Model based Neural Vocoder with Adaptive Noise Spectral Shaping. 803-807 - Sangjun Park, Kihyun Choo, Joohyung Lee, Anton V. Porov, Konstantin Osipov, June Sig Sung:
Bunched LPCNet2: Efficient Neural Vocoders Covering Devices from Cloud to Edge. 808-812 - Jae-Sung Bae, Jinhyeok Yang, Taejun Bak, Young-Sun Joo:
Hierarchical and Multi-Scale Variational Autoencoder for Diverse and Natural Non-Autoregressive Text-to-Speech. 813-817 - Krishna Subramani, Jean-Marc Valin, Umut Isik, Paris Smaragdis, Arvindh Krishnaswamy:
End-to-end LPCNet: A Neural Vocoder With Fully-Differentiable LPC Estimation. 818-822 - Perry Lam, Huayun Zhang, Nancy F. Chen, Berrak Sisman:
EPIC TTS Models: Empirical Pruning Investigations Characterizing Text-To-Speech Models. 823-827 - Karolos Nikitaras, Georgios Vamvoukakis, Nikolaos Ellinas, Konstantinos Klapsas, Konstantinos Markopoulos, Spyros Raptis, June Sig Sung, Gunu Jho, Aimilios Chalamandaris, Pirros Tsiakoulis:
Fine-grained Noise Control for Multispeaker Speech Synthesis. 828-832 - Hubert Siuzdak, Piotr Dura, Pol van Rijn, Nori Jacoby:
WavThruVec: Latent speech representation as intermediate features for neural speech synthesis. 833-837 - Ivan Vovk, Tasnima Sadekova, Vladimir Gogoryan, Vadim Popov, Mikhail A. Kudinov, Jiansheng Wei:
Fast Grad-TTS: Towards Efficient Diffusion-Based Speech Generation on CPU. 838-842 - Alexander H. Liu, Cheng-I Lai, Wei-Ning Hsu, Michael Auli, Alexei Baevski, James R. Glass:
Simple and Effective Unsupervised Speech Synthesis. 843-847 - Reo Yoneyama, Yi-Chiao Wu, Tomoki Toda:
Unified Source-Filter GAN with Harmonic-plus-Noise Source Excitation Generation. 848-852
Show and Tell I
- Taejin Park, Nithin Rao Koluguri, Fei Jia, Jagadeesh Balam, Boris Ginsburg:
NeMo Open Source Speaker Diarization System. 853-854 - Baihan Lin:
Voice2Alliance: Automatic Speaker Diarization and Quality Assurance of Conversational Alignment. 855-856 - Rishabh Kumar, Devaraja Adiga, Mayank Kothyari, Jatin Dalal, Ganesh Ramakrishnan, Preethi Jyothi:
VAgyojaka: An Annotating and Post-Editing Tool for Automatic Speech Recognition. 857-858 - Alzahra Badi, Chungho Park, Min-Seok Keum, Miguel Alba, Youngsuk Ryu, Jeongmin Bae:
SKYE: More than a conversational AI. 859-860
Spatial Audio
- Hokuto Munakata, Ryu Takeda, Kazunori Komatani:
Training Data Generation with DOA-based Selecting and Remixing for Unsupervised Training of Deep Separation Models. 861-865 - Hangting Chen, Yi Yang, Feng Dang, Pengyuan Zhang:
Beam-Guided TasNet: An Iterative Speech Separation Framework with Multi-Channel Output. 866-870 - Feifei Xiong, Pengyu Wang, Zhongfu Ye, Jinwei Feng:
Joint Estimation of Direction-of-Arrival and Distance for Arrays with Directional Sensors based on Sparse Bayesian Learning. 871-875 - Ho-Hsiang Wu, Magdalena Fuentes, Prem Seetharaman, Juan Pablo Bello:
How to Listen? Rethinking Visual Sound Localization. 876-880 - Zhiheng Ouyang, Miao Wang, Wei-Ping Zhu:
Small Footprint Neural Networks for Acoustic Direction of Arrival Estimation. 881-885 - Xiaoyu Wang, Xiangyu Kong, Xiulian Peng, Yan Lu:
Multi-Modal Multi-Correlation Learning for Audio-Visual Speech Separation. 886-890 - Haoran Yin, Meng Ge, Yanjie Fu, Gaoyan Zhang, Longbiao Wang, Lei Zhang, Lin Qiu, Jianwu Dang:
MIMO-DoAnet: Multi-channel Input and Multiple Outputs DoA Network with Unknown Number of Sound Sources. 891-895 - Yanjie Fu, Meng Ge, Haoran Yin, Xinyuan Qian, Longbiao Wang, Gaoyan Zhang, Jianwu Dang:
Iterative Sound Source Localization for Unknown Number of Sources. 896-900 - Katharine Patterson, Kevin W. Wilson, Scott Wisdom, John R. Hershey:
Distance-Based Sound Separation. 901-905 - Junjie Li, Meng Ge, Zexu Pan, Longbiao Wang, Jianwu Dang:
VCSE: Time-Domain Visual-Contextual Speaker Extraction Network. 906-910 - Ali Aroudi, Stefan Uhlich, Marc Ferras Font:
TRUNet: Transformer-Recurrent-U Network for Multi-channel Reverberant Sound Source Separation. 911-915
Single-channel Speech Enhancement II
- Xiaofeng Ge, Jiangyu Han, Yanhua Long, Haixin Guan:
PercepNet+: A Phase and SNR Aware PercepNet for Real-Time Speech Enhancement. 916-920 - Zhuangqi Chen, Pingjian Zhang:
Lightweight Full-band and Sub-band Fusion Network for Real Time Speech Enhancement. 921-925 - Jiaming Cheng, Ruiyu Liang, Yue Xie, Li Zhao, Björn W. Schuller, Jie Jia, Yiyuan Peng:
Cross-Layer Similarity Knowledge Distillation for Speech Enhancement. 926-930 - Feifei Xiong, Weiguang Chen, Pengyu Wang, Xiaofei Li, Jinwei Feng:
Spectro-Temporal SubNet for Real-Time Monaural Speech Denoising and Dereverberation. 931-935 - Ruizhe Cao, Sherif Abdulatif, Bin Yang:
CMGAN: Conformer-based Metric GAN for Speech Enhancement. 936-940 - Zeyuan Wei, Li Hao, Xueliang Zhang:
Model Compression by Iterative Pruning with Knowledge Distillation and Its Application to Speech Enhancement. 941-945 - Chenhui Zhang, Xiang Pan:
Single-channel speech enhancement using Graph Fourier Transform. 946-950 - Zilu Guo, Xu Xu, Zhongfu Ye:
Joint Optimization of the Module and Sign of the Spectral Real Part Based on CRN for Speech Denoising. 951-955 - Hao Zhang, Ashutosh Pandey, DeLiang Wang:
Attentive Recurrent Network for Low-Latency Active Noise Control. 956-960 - Jen-Hung Huang, Chung-Hsien Wu:
Memory-Efficient Multi-Step Speech Enhancement with Neural ODE. 961-965 - Xinmeng Xu, Yang Wang, Jie Jia, Binbin Chen, Jianjun Hao:
GLD-Net: Improving Monaural Speech Enhancement by Learning Global and Local Dependency Features with GLD Block. 966-970 - Xinmeng Xu, Yang Wang, Jie Jia, Binbin Chen, Dejun Li:
Improving Visual Speech Enhancement Network by Learning Audio-visual Affinity with Multi-head Attention. 971-975 - Jun Chen, Wei Rao, Zilin Wang, Zhiyong Wu, Yannan Wang, Tao Yu, Shidong Shang, Helen Meng:
Speech Enhancement with Fullband-Subband Cross-Attention Network. 976-980 - Cheng Yu, Szu-Wei Fu, Tsun-An Hsieh, Yu Tsao, Mirco Ravanelli:
OSSEM: one-shot speaker adaptive speech enhancement using meta learning. 981-985 - Wenbin Jiang, Tao Liu, Kai Yu:
Efficient Speech Enhancement with Neural Homomorphic Synthesis. 986-990 - Manthan Thakker, Sefik Emre Eskimez, Takuya Yoshioka, Huaming Wang:
Fast Real-time Personalized Speech Enhancement: End-to-End Enhancement Network (E3Net) and Knowledge Distillation. 991-995 - Hiroshi Sato, Tsubasa Ochiai, Marc Delcroix, Keisuke Kinoshita, Takafumi Moriya, Naoki Makishima, Mana Ihori, Tomohiro Tanaka, Ryo Masumura:
Strategies to Improve Robustness of Target Speech Extraction to Enrollment Variations. 996-1000
Novel Models and Training Methods for ASR II
- Haaris Mehmood, Agnieszka Dobrowolska, Karthikeyan Saravanan, Mete Ozay:
FedNST: Federated Noisy Student Training for Automatic Speech Recognition. 1001-1005 - Li Fu, Xiaoxiao Li, Runyu Wang, Lu Fan, Zhengchen Zhang, Meng Chen, Youzheng Wu, Xiaodong He:
SCaLa: Supervised Contrastive Learning for End-to-End Speech Recognition. 1006-1010 - Yukun Liu, Ta Li, Pengyuan Zhang, Yonghong Yan:
NAS-SCAE: Searching Compact Attention-based Encoders For End-to-end Automatic Speech Recognition. 1011-1015 - Kun Wei, Yike Zhang, Sining Sun, Lei Xie, Long Ma:
Leveraging Acoustic Contextual Representation by Audio-textual Cross-modal Learning for Conversational ASR. 1016-1020 - Guodong Ma, Pengfei Hu, Nurmemet Yolwas, Shen Huang, Hao Huang:
PM-MMUT: Boosted Phone-mask Data Augmentation using Multi-Modeling Unit Training for Phonetic-Reduction-Robust E2E Speech Recognition. 1021-1025 - Kartik Audhkhasi, Yinghui Huang, Bhuvana Ramabhadran, Pedro J. Moreno:
Analysis of Self-Attention Head Diversity for Conformer-based Automatic Speech Recognition. 1026-1030 - Weiran Wang, Tongzhou Chen, Tara N. Sainath, Ehsan Variani, Rohit Prabhavalkar, W. Ronny Huang, Bhuvana Ramabhadran, Neeraj Gaur, Sepand Mavandadi, Cal Peyser, Trevor Strohman, Yanzhang He, David Rybach:
Improving Rare Word Recognition with LM-aware MWER Training. 1031-1035 - Mohammad Zeineldeen, Jingjing Xu, Christoph Lüscher, Ralf Schlüter, Hermann Ney:
Improving the Training Recipe for a Robust Conformer-based Hybrid Model. 1036-1040 - Aleksandr Laptev, Somshubra Majumdar, Boris Ginsburg:
CTC Variations Through New WFST Topologies. 1041-1045 - Martin Sustek, Samik Sadhu, Hynek Hermansky:
Dealing with Unknowns in Continual Learning for End-to-end Automatic Speech Recognition. 1046-1050 - Chenfeng Miao, Kun Zou, Ziyang Zhuang, Tao Wei, Jun Ma, Shaojun Wang, Jing Xiao:
Towards Efficiently Learning Monotonic Alignments for Attention-based End-to-End Speech Recognition. 1051-1055 - Jisi Zhang, Catalin Zorila, Rama Doddipatla, Jon Barker:
On monoaural speech enhancement for automatic recognition of real noisy speech using mixture invariant training. 1056-1060 - Selen Hande Kabil, Hervé Bourlard:
From Undercomplete to Sparse Overcomplete Autoencoders to Improve LF-MMI based Speech Recognition. 1061-1065 - Tomohiro Tanaka, Ryo Masumura, Hiroshi Sato, Mana Ihori, Kohei Matsuura, Takanori Ashihara, Takafumi Moriya:
Domain Adversarial Self-Supervised Speech Representation Learning for Improving Unknown Domain Downstream Tasks. 1066-1070 - Takashi Maekaku, Yuya Fujita, Yifan Peng, Shinji Watanabe:
Attention Weight Smoothing Using Prior Distributions for Transformer-Based End-to-End ASR. 1071-1075
Spoken Dialogue Systems and Multimodality
- Naokazu Uchida, Takeshi Homma, Makoto Iwayama, Yasuhiro Sogawa:
Reducing Offensive Replies in Open Domain Dialogue Systems. 1076-1080 - Ting-Wei Wu, Biing-Hwang Juang:
Induce Spoken Dialog Intents via Deep Unsupervised Context Contrastive Clustering. 1081-1085 - Fumio Nihei, Ryo Ishii, Yukiko I. Nakano, Kyosuke Nishida, Ryo Masumura, Atsushi Fukayama, Takao Nakamura:
Dialogue Acts Aided Important Utterance Detection Based on Multiparty and Multimodal Information. 1086-1090 - Dhanush Bekal, Sundararajan Srinivasan, Srikanth Ronanki, Sravan Bodapati, Katrin Kirchhoff:
Contextual Acoustic Barge-In Classification for Spoken Dialog Systems. 1091-1095 - Peilin Zhou, Dading Chong, Helin Wang, Qingcheng Zeng:
Calibrate and Refine! A Novel and Agile Framework for ASR Error Robust Intent Detection. 1096-1100 - Lingyun Feng, Jianwei Yu, Yan Wang, Songxiang Liu, Deng Cai, Haitao Zheng:
ASR-Robust Natural Language Understanding on ASR-GLUE dataset. 1101-1105 - Mai Hoang Dao, Thinh Hung Truong, Dat Quoc Nguyen:
From Disfluency Detection to Intent Detection and Slot Filling. 1106-1110 - Hengshun Zhou, Jun Du, Gongzhen Zou, Zhaoxu Nian, Chin-Hui Lee, Sabato Marco Siniscalchi, Shinji Watanabe, Odette Scharenborg, Jingdong Chen, Shifu Xiong, Jianqing Gao:
Audio-Visual Wake Word Spotting in MISP2021 Challenge: Dataset Release and Deep Analysis. 1111-1115 - Christina Sartzetaki, Georgios Paraskevopoulos, Alexandros Potamianos:
Extending Compositional Attention Networks for Social Reasoning in Videos. 1116-1120 - Shiquan Wang, Yuke Si, Xiao Wei, Longbiao Wang, Zhiqiang Zhuang, Xiaowang Zhang, Jianwu Dang:
TopicKS: Topic-driven Knowledge Selection for Knowledge-grounded Dialogue Generation. 1121-1125 - Andreas Liesenfeld, Mark Dingemanse:
Bottom-up discovery of structure and variation in response tokens ('backchannels') across diverse languages. 1126-1130 - Yi Zhu, Zexun Wang, Hang Liu, Peiying Wang, Mingchao Feng, Meng Chen, Xiaodong He:
Cross-modal Transfer Learning via Multi-grained Alignment for End-to-End Spoken Language Understanding. 1131-1135 - Keiko Ochi, Nobutaka Ono, Keiho Owada, Miho Kuroda, Shigeki Sagayama, Hidenori Yamasue:
Use of Nods Less Synchronized with Turn-Taking and Prosody During Conversations in Adults with Autism. 1136-1140
Show and Tell I(VR)
- Denis Ivanko, Dmitry Ryumin, Alexey M. Kashevnik, Alexandr Axyonov, Andrey Kitenko, Igor Lashkov, Alexey Karpov:
DAVIS: Driver's Audio-Visual Speech recognition. 1141-1142
Speech Emotion Recognition I
- Einari Vaaras, Manu Airaksinen, Okko Räsänen:
Analysis of Self-Supervised Learning and Dimensionality Reduction Methods in Clustering-Based Active Learning for Speech Emotion Recognition. 1143-1147 - Chun-Yu Chen, Yun-Shao Lin, Chi-Chun Lee:
Emotion-Shift Aware CRF for Decoding Emotion Sequence in Conversation. 1148-1152 - Bo-Hao Su, Chi-Chun Lee:
Vaccinating SER to Neutralize Adversarial Attacks with Self-Supervised Augmentation Strategy. 1153-1157 - Jack Parry, Eric DeMattos, Anita Klementiev, Axel Ind, Daniela Morse-Kopp, Georgia Clarke, Dimitri Palaz:
Speech Emotion Recognition in the Wild using Multi-task and Adversarial Learning. 1158-1162 - Ashishkumar Prabhakar Gudmalwar, Biplove Basel, Anirban Dutta, Ch V. Rama Rao:
The Magnitude and Phase based Speech Representation Learning using Autoencoder for Classifying Speech Emotions using Deep Canonical Correlation Analysis. 1163-1167 - Lucas Goncalves, Carlos Busso:
Improving Speech Emotion Recognition Using Self-Supervised Learning with Domain-Specific Audiovisual Tasks. 1168-1172
Single-channel Speech Enhancement I
- Yuma Koizumi, Shigeki Karita, Arun Narayanan, Sankaran Panchapagesan, Michiel Bacchiani:
SNRi Target Training for Joint Speech Enhancement and Recognition. 1173-1177 - Yutaro Sanada, Takumi Nakagawa, Yuichiro Wada, Kosaku Takanashi, Yuhui Zhang, Kiichi Tokuyama, Takafumi Kanamori, Tomonori Yamada:
Deep Self-Supervised Learning of Speech Denoising from Noisy Speeches. 1178-1182 - Chi-Chang Lee, Cheng-Hung Hu, Yu-Chen Lin, Chu-Song Chen, Hsin-Min Wang, Yu Tsao:
NASTAR: Noise Adaptive Speech Enhancement with Target-Conditional Resampling. 1183-1187 - Ivan Shchekotov, Pavel K. Andreev, Oleg Ivanov, Aibek Alanov, Dmitry P. Vetrov:
FFC-SE: Fast Fourier Convolution for Speech Enhancement. 1188-1192 - Or Tal, Moshe Mandel, Felix Kreuk, Yossi Adi:
A Systematic Comparison of Phonetic Aware Techniques for Speech Enhancement. 1193-1197 - Wooseok Shin, Hyun Joon Park, Jin Sob Kim, Byung Hoon Lee, Sung Won Han:
Multi-View Attention Transfer for Efficient Speech Enhancement. 1198-1202
Speech Synthesis: New Applications
- Nabarun Goswami, Tatsuya Harada:
SATTS: Speaker Attractor Text to Speech, Learning to Speak by Learning to Separate. 1203-1207 - Talia Ben Simon, Felix Kreuk, Faten Awwad, Jacob T. Cohen, Joseph Keshet:
Correcting Mispronunciations in Speech using Spectrogram Inpainting. 1208-1212 - Jason Fong, Daniel Lyth, Gustav Eje Henter, Hao Tang, Simon King:
Speech Audio Corrector: using speech from non-target speakers for one-off correction of mispronunciations in grapheme-input text-to-speech. 1213-1217 - Wen-Chin Huang, Dejan Markovic, Alexander Richard, Israel Dejene Gebru, Anjali Menon:
End-to-End Binaural Speech Synthesis. 1218-1222 - Julia Koch, Florian Lux, Nadja Schauffler, Toni Bernhart, Felix Dieterle, Jonas Kuhn, Sandra Richter, Gabriel Viehhauser, Ngoc Thang Vu:
PoeticTTS - Controllable Poetry Reading for Literary Studies. 1223-1227 - Paul Konstantin Krug, Peter Birkholz, Branislav Gerazov, Daniel Rudolph van Niekerk, Anqi Xu, Yi Xu:
Articulatory Synthesis for Data Augmentation in Phoneme Recognition. 1228-1232
Spoken Language Understanding I
- Jihyun Lee, Gary Geunbae Lee:
SF-DST: Few-Shot Self-Feeding Reading Comprehension Dialogue State Tracking with Auxiliary Task. 1233-1237 - Oralie Cattan, Sahar Ghannay, Christophe Servan, Sophie Rosset:
Benchmarking Transformers-based models on French Spoken Language Understanding tasks. 1238-1242 - Seong-Hwan Heo, WonKee Lee, Jong-Hyeok Lee:
mcBERT: Momentum Contrastive Learning with BERT for Zero-Shot Slot Filling. 1243-1247 - Pu Wang, Hugo Van hamme:
Bottleneck Low-rank Transformers for Low-resource Spoken Language Understanding. 1248-1252 - Anirudh Raju, Milind Rao, Gautam Tiwari, Pranav Dheram, Bryan Anderson, Zhe Zhang, Chul Lee, Bach Bui, Ariya Rastrow:
On joint training with interfaces for spoken language understanding. 1253-1257 - Vineet Garg, Ognjen Rudovic, Pranay Dighe, Ahmed Hussen Abdelaziz, Erik Marchi, Saurabh Adya, Chandra Dhir, Ahmed H. Tewfik:
Device-Directed Speech Detection: Regularization via Distillation for Weakly-Supervised Models. 1258-1262
Inclusive and Fair Speech Technologies I
- Perez Ogayo, Graham Neubig, Alan W. Black:
Building African Voices. 1263-1267 - Pranav Dheram, Murugesan Ramakrishnan, Anirudh Raju, I-Fan Chen, Brian King, Katherine Powell, Melissa Saboowala, Karan Shetty, Andreas Stolcke:
Toward Fairness in Speech Recognition: Discovery and mitigation of performance disparities. 1268-1272 - May Pik Yu Chan, June Choe, Aini Li, Yiran Chen, Xin Gao, Nicole R. Holliday:
Training and typological bias in ASR performance for world Englishes. 1273-1277
Inclusive and Fair Speech Technologies II
- Marcely Zanon Boito, Laurent Besacier, Natalia A. Tomashenko, Yannick Estève:
A Study of Gender Impact in Self-supervised Models for Speech-to-Text Systems. 1278-1282 - Alexander Johnson, Kevin Everson, Vijay Ravi, Anissa Gladney, Mari Ostendorf, Abeer Alwan:
Automatic Dialect Density Estimation for African American English. 1283-1287 - Kunnar Kukk, Tanel Alumäe:
Improving Language Identification of Accented Speech. 1288-1292 - Wiebke Toussaint, Lauriane Gorce, Aaron Yi Ding:
Design Guidelines for Inclusive Speaker Verification Evaluation Datasets. 1293-1297 - Viet Anh Trinh, Pegah Ghahremani, Brian John King, Jasha Droppo, Andreas Stolcke, Roland Maas:
Reducing Geographic Disparities in Automatic Speech Recognition via Elastic Weight Consolidation. 1298-1302
Phonetics I
- Takuya Kunihara, Chuanbo Zhu, Nobuaki Minematsu, Noriko Nakanishi:
Gradual Improvements Observed in Learners' Perception and Production of L2 Sounds Through Continuing Shadowing Practices on a Daily Basis. 1303-1307 - Christin Kirchhübel, Georgina Brown:
Spoofed speech from the perspective of a forensic phonetician. 1308-1312 - Hae-Sung Jeon, Stephen Nichols:
Investigating Prosodic Variation in British English Varieties using ProPer. 1313-1317 - Hyun Kyung Hwang, Manami Hirayama, Takaomi Kato:
Perceived prominence and downstep in Japanese. 1318-1321 - Andrea Alicehajic, Silke Hamann:
The discrimination of [zi]-[dʑi] by Japanese listeners and the prospective phonologization of /zi/. 1322-1326 - Ingo Langheinrich, Simon Stone, Xinyu Zhang, Peter Birkholz:
Glottal inverse filtering based on articulatory synthesis and deep learning. 1327-1331 - Bogdan Ludusan, Marin Schröer, Petra Wagner:
Investigating phonetic convergence of laughter in conversation. 1332-1336 - Véronique Delvaux, Audrey Lavallée, Fanny Degouis, Xavier Saloppe, Jean-Louis Nandrino, Thierry Pham:
Telling self-defining memories: An acoustic study of natural emotional speech productions. 1337-1341 - Laura Spinu, Ioana Vasilescu, Lori Lamel, Jason Lilley:
Voicing neutralization in Romanian fricatives across different speech styles. 1342-1346 - Sishi Liao, Phil Hoole, Conceição Cunha, Esther Kunay, Aletheia Cui, Lia Saki Bucar Shigemori, Felicitas Kleber, Dirk Voit, Jens Frahm, Jonathan Harrington:
Nasal Coda Loss in the Chengdu Dialect of Mandarin: Evidence from RT-MRI. 1347-1351 - Philipp Buech, Simon Roessig, Lena Pagel, Doris Mücke, Anne Hermes:
ema2wav: doing articulation by Praat. 1352-1356
Multi-, Cross-lingual and Other Topics in ASR I
- Lars Rumberg, Christopher Gebauer, Hanna Ehlert, Ulrike Lüdtke, Jörn Ostermann:
Improving Phonetic Transcriptions of Children's Speech by Pronunciation Modelling with Constrained CTC-Decoding. 1357-1361 - Soky Kak, Sheng Li, Masato Mimura, Chenhui Chu, Tatsuya Kawahara:
Leveraging Simultaneous Translation for Enhancing Transcription of Low-resource Language via Cross Attention Mechanism. 1362-1366 - Saida Mussakhojayeva, Yerbolat Khassanov, Huseyin Atakan Varol:
KSC2: An Industrial-Scale Open-Source Kazakh Speech Corpus. 1367-1371 - Tünde Szalay, Mostafa Ali Shahin, Beena Ahmed, Kirrie J. Ballard:
Knowledge of accent differences can be used to predict speech recognition. 1372-1376 - Maximilian Karl Scharf, Sabine Hochmuth, Lena L. N. Wong, Birger Kollmeier, Anna Warzybok:
Lombard Effect for Bilingual Speakers in Cantonese and English: importance of spectro-temporal features. 1377-1381 - Martin Flechl, Shou-Chun Yin, Junho Park, Peter Skala:
End-to-end speech recognition modeling from de-identified data. 1382-1386 - Aditya Yadavalli, Mirishkar Sai Ganesh, Anil Kumar Vuppala:
Multi-Task End-to-End Model for Telugu Dialect and Speech Recognition. 1387-1391 - Jiamin Xie, John H. L. Hansen:
DEFORMER: Coupling Deformed Localized Patterns with Global Context for Robust End-to-end Speech Recognition. 1392-1396
Zero, low-resource and multi-modal speech recognition I
- Yuna Lee, Seung Jun Baek:
Keyword Spotting with Synthetic Data using Heterogeneous Knowledge Distillation. 1397-1401 - Maureen de Seyssel, Marvin Lavechin, Yossi Adi, Emmanuel Dupoux, Guillaume Wisniewski:
Probing phoneme, language and speaker information in unsupervised speech representations. 1402-1406 - Andrei Bîrladeanu, Helen Minnis, Alessandro Vinciarelli:
Automatic Detection of Reactive Attachment Disorder Through Turn-Taking Analysis in Clinical Child-Caregiver Sessions. 1407-1410 - Eesung Kim, Jae-Jin Jeon, Hyeji Seo, Hoon Kim:
Automatic Pronunciation Assessment using Self-Supervised Speech Representation Learning. 1411-1415 - Tyler Miller, David Harwath:
Exploring Few-Shot Fine-Tuning Strategies for Models of Visually Grounded Speech. 1416-1420 - Dongseong Hwang, Khe Chai Sim, Zhouyuan Huo, Trevor Strohman:
Pseudo Label Is Better Than Human Label. 1421-1425 - Werner van der Merwe, Herman Kamper, Johan Adam du Preez:
A Temporal Extension of Latent Dirichlet Allocation for Unsupervised Acoustic Unit Discovery. 1426-1430
Speaker Embedding and Diarization
- Siqi Zheng, Hongbin Suo, Qian Chen:
PRISM: Pre-trained Indeterminate Speaker Representation Model for Speaker Diarization and Speaker Verification. 1431-1435 - Xiaoyi Qin, Na Li, Chao Weng, Dan Su, Ming Li:
Cross-Age Speaker Verification: Learning Age-Invariant Speaker Embeddings. 1436-1440 - Weiqing Wang, Ming Li, Qingjian Lin:
Online Target Speaker Voice Activity Detection for Speaker Diarization. 1441-1445 - Niko Brummer, Albert Swart, Ladislav Mosner, Anna Silnova, Oldrich Plchot, Themos Stafylakis, Lukás Burget:
Probabilistic Spherical Discriminant Analysis: An Alternative to PLDA for length-normalized embeddings. 1446-1450 - Bin Gu:
Deep speaker embedding with frame-constrained training strategy for speaker verification. 1451-1455 - Yifan Chen, Yifan Guo, Qingxuan Li, Gaofeng Cheng, Pengyuan Zhang, Yonghong Yan:
Interrelate Training and Searching: A Unified Online Clustering Framework for Speaker Diarization. 1456-1460 - Mao-Kui He, Jun Du, Chin-Hui Lee:
End-to-End Audio-Visual Neural Speaker Diarization. 1461-1465 - Yanyan Yue, Jun Du, Mao-Kui He, Yu Ting Yeung, Renyu Wang:
Online Speaker Diarization with Core Samples Selection. 1466-1470 - Chenyu Yang, Yu Wang:
Robust End-to-end Speaker Diarization with Generic Neural Clustering. 1471-1475 - Tao Liu, Shuai Fan, Xu Xiang, Hongbo Song, Shaoxiong Lin, Jiaqi Sun, Tianyuan Han, Siyuan Chen, Binwei Yao, Sen Liu, Yifei Wu, Yanmin Qian, Kai Yu:
MSDWild: Multi-modal Speaker Diarization Dataset in the Wild. 1476-1480 - Md. Iftekhar Tanveer, Diego Casabuena, Jussi Karlgren, Rosie Jones:
Unsupervised Speaker Diarization that is Agnostic to Language, Overlap-Aware, and Tuning Free. 1481-1485 - Keisuke Kinoshita, Thilo von Neumann, Marc Delcroix, Christoph Böddeker, Reinhold Haeb-Umbach:
Utterance-by-utterance overlap-aware neural diarization with Graph-PIT. 1486-1490 - Jie Wang, Yuji Liu, Binling Wang, Yiming Zhi, Song Li, Shipeng Xia, Jiayang Zhang, Feng Tong, Lin Li, Qingyang Hong:
Spatial-aware Speaker Diarizaiton for Multi-channel Multi-party Meeting. 1491-1495
Acoustic Event Detection and Classification
- Yunhao Liang, Yanhua Long, Yijie Li, Jiaen Liang:
Selective Pseudo-labeling and Class-wise Discriminative Fusion for Sound Event Detection. 1496-1500 - Peng Liu, Songbin Li, Jigang Tang:
An End-to-End Macaque Voiceprint Verification Method Based on Channel Fusion Mechanism. 1501-1505 - Liang Xu, Jing Wang, Lizhong Wang, Sijun Bi, Jianqian Zhang, Qiuyue Ma:
Human Sound Classification based on Feature Fusion Method with Air and Bone Conducted Signal. 1506-1510 - Dongchao Yang, Helin Wang, Zhongjie Ye, Yuexian Zou, Wenwu Wang:
RaDur: A Reference-aware and Duration-robust Network for Target Sound Detection. 1511-1515 - Achyut Mani Tripathi, Konark Paul:
Temporal Self Attention-Based Residual Network for Environmental Sound Classification. 1516-1520 - Juncheng Li, Shuhui Qu, Po-Yao Huang, Florian Metze:
AudioTagging Done Right: 2nd comparison of deep learning methods for environmental sound classification. 1521-1525 - Helin Wang, Dongchao Yang, Chao Weng, Jianwei Yu, Yuexian Zou:
Improving Target Sound Extraction with Timestamp Information. 1526-1530 - Ying Hu, Xiujuan Zhu, Yunlong Li, Hao Huang, Liang He:
A Multi-grained based Attention Network for Semi-supervised Sound Event Detection. 1531-1535 - Sangwook Park, Sandeep Reddy Kothinti, Mounya Elhilali:
Temporal coding with magnitude-phase regularization for sound event detection. 1536-1540 - Nian Shao, Erfan Loweimi, Xiaofei Li:
RCT: Random consistency training for semi-supervised sound event detection. 1541-1545 - Yifei Xin, Dongchao Yang, Yuexian Zou:
Audio Pyramid Transformer with Domain Adaption for Weakly Supervised Sound Event Detection and Audio Classification. 1546-1550 - Yu Wang, Mark Cartwright, Juan Pablo Bello:
Active Few-Shot Learning for Sound Event Detection. 1551-1555 - Tong Ye, Shijing Si, Jianzong Wang, Ning Cheng, Jing Xiao:
Uncertainty Calibration for Deep Audio Classifiers. 1556-1560 - Yuanbo Hou, Dick Botteldooren:
Event-related data conditioning for acoustic event classification. 1561-1565
Speech Synthesis: Acoustic Modeling and Neural Waveform Generation II
- Haohan Guo, Hui Lu, Xixin Wu, Helen Meng:
A Multi-Scale Time-Frequency Spectrogram Discriminator for GAN-based Non-Autoregressive TTS. 1566-1570 - Dacheng Yin, Chuanxin Tang, Yanqing Liu, Xiaoqiang Wang, Zhiyuan Zhao, Yucheng Zhao, Zhiwei Xiong, Sheng Zhao, Chong Luo:
RetrieverTTS: Modeling Decomposed Factors for Text-Based Speech Insertion. 1571-1575 - Manh Luong, Viet-Anh Tran:
FlowVocoder: A small Footprint Neural Vocoder based Normalizing Flow for Speech Synthesis. 1576-1580 - Yanqing Liu, Ruiqing Xue, Lei He, Xu Tan, Sheng Zhao:
DelightfulTTS 2: End-to-End Speech Synthesis with Adversarial Vector-Quantized Auto-Encoders. 1581-1585 - Xin Yuan, Robin Feng, Mingming Ye, Cheng Tuo, Minghang Zhang:
AdaVocoder: Adaptive Vocoder for Custom Voice. 1586-1590 - Shengyuan Xu, Wenxiao Zhao, Jing Guo:
RefineGAN: Universally Generating Waveform Better than Ground Truth with Highly Accurate Pitch and Intensity Responses. 1591-1595 - Chenpeng Du, Yiwei Guo, Xie Chen, Kai Yu:
VQTTS: High-Fidelity Text-to-Speech Synthesis with Self-Supervised VQ Acoustic Feature. 1596-1600 - Mengnan He, Tingwei Guo, Zhenxing Lu, Ruixiong Zhang, Caixia Gong:
Improving GAN-based vocoder for fast and high-quality speech synthesis. 1601-1605 - Yuanhao Yi, Lei He, Shifeng Pan, Xi Wang, Yuchao Zhang:
SoftSpeech: Unsupervised Duration Model in FastSpeech 2. 1606-1610 - Haohan Guo, Feng-Long Xie, Frank K. Soong, Xixin Wu, Helen Meng:
A Multi-Stage Multi-Codebook VQ-VAE Approach to High-Performance Neural TTS. 1611-1615 - Yuhan Li, Ying Shen, Dongqing Wang, Lin Zhang:
SiD-WaveFlow: A Low-Resource Vocoder Independent of Prior Knowledge. 1616-1620 - Takeru Gorai, Daisuke Saito, Nobuaki Minematsu:
Text-to-speech synthesis using spectral modeling based on non-negative autoencoder. 1621-1625 - Hiroki Kanagawa, Yusuke Ijima, Hiroyuki Toda:
Joint Modeling of Multi-Sample and Subband Signals for Fast Neural Vocoding on CPU. 1626-1630 - Takuhiro Kaneko, Hirokazu Kameoka, Kou Tanaka, Shogo Seki:
MISRNet: Lightweight Neural Vocoder Using Multi-Input Single Shared Residual Blocks. 1631-1635 - Chenfeng Miao, Ting Chen, Minchuan Chen, Jun Ma, Shaojun Wang, Jing Xiao:
A compact transformer-based GAN vocoder. 1636-1640 - Hideyuki Tachibana, Muneyoshi Inahara, Mocho Go, Yotaro Katayama, Yotaro Watanabe:
Diffusion Generative Vocoder for Fullband Speech Synthesis Based on Weak Third-order SDE Solver. 1641-1645
ASR: Architecture and Search
- Ehsan Variani, Michael Riley, David Rybach, Cyril Allauzen, Tongzhou Chen, Bhuvana Ramabhadran:
On Adaptive Weight Interpolation of the Hybrid Autoregressive Transducer. 1646-1650 - Ting-Wei Wu, I-Fan Chen, Ankur Gandhe:
Learning to rank with BERT-based confidence models in ASR rescoring. 1651-1655 - Jiatong Shi, George Saon, David Haws, Shinji Watanabe, Brian Kingsbury:
VQ-T: RNN Transducers using Vector-Quantized Prediction Network States. 1656-1660 - Binbin Zhang, Di Wu, Zhendong Peng, Xingchen Song, Zhuoyuan Yao, Hang Lv, Lei Xie, Chao Yang, Fuping Pan, Jianwei Niu:
WeNet 2.0: More Productive End-to-End Speech Recognition Toolkit. 1661-1665 - Yufei Liu, Rao Ma, Haihua Xu, Yi He, Zejun Ma, Weibin Zhang:
Internal Language Model Estimation Through Explicit Context Vector Learning for Attention-based Encoder-decoder ASR. 1666-1670 - Zehan Li, Haoran Miao, Keqi Deng, Gaofeng Cheng, Sanli Tian, Ta Li, Yonghong Yan:
Improving Streaming End-to-End ASR on Transformer-based Causal Models with Encoder States Revision Strategies. 1671-1675 - Ye Bai, Jie Li, Wenjing Han, Hao Ni, Kaituo Xu, Zhuo Zhang, Cheng Yi, Xiaorui Wang:
Parameter-Efficient Conformers via Sharing Sparsely-Gated Experts for End-to-End Speech Recognition. 1676-1680 - Zhanheng Yang, Sining Sun, Jin Li, Xiaoming Zhang, Xiong Wang, Long Ma, Lei Xie:
CaTT-KWS: A Multi-stage Customized Keyword Spotting Framework based on Cascaded Transducer-Transformer. 1681-1685 - Rui Wang, Qibing Bai, Junyi Ao, Long Zhou, Zhixiang Xiong, Zhihua Wei, Yu Zhang, Tom Ko, Haizhou Li:
LightHuBERT: Lightweight and Configurable Speech Representation Learning with Once-for-All Hidden-Unit BERT. 1686-1690 - Jash Rathod, Nauman Dawalatabad, Shatrughan Singh, Dhananjaya Gowda:
Multi-stage Progressive Compression of Conformer Transducer for On-device Speech Recognition. 1691-1695 - Weiran Wang, Ke Hu, Tara N. Sainath:
Streaming Align-Refine for Non-autoregressive Deliberation. 1696-1700 - Rongmei Lin, Yonghui Xiao, Tien-Ju Yang, Ding Zhao, Li Xiong, Giovanni Motta, Françoise Beaufays:
Federated Pruning: Improving Neural Network Efficiency with Federated Learning. 1701-1705 - Shaojin Ding, Weiran Wang, Ding Zhao, Tara N. Sainath, Yanzhang He, Robert David, Rami Botros, Xin Wang, Rina Panigrahy, Qiao Liang, Dongseong Hwang, Ian McGraw, Rohit Prabhavalkar, Trevor Strohman:
A Unified Cascaded Encoder ASR Model for Dynamic Model Sizes. 1706-1710 - Shaojin Ding, Phoenix Meadowlark, Yanzhang He, Lukasz Lew, Shivani Agrawal, Oleg Rybakov:
4-bit Conformer with Native Quantization Aware Training for Speech Recognition. 1711-1715 - Qiang Xu, Tongtong Song, Longbiao Wang, Hao Shi, Yuqin Lin, Yongjie Lv, Meng Ge, Qiang Yu, Jianwu Dang:
Self-Distillation Based on High-level Information Supervision for Compressing End-to-End ASR Model. 1716-1720
Spoken Language Processing II
- Ye Jia, Yifan Ding, Ankur Bapna, Colin Cherry, Yu Zhang, Alexis Conneau, Nobu Morioka:
Leveraging unsupervised and weakly-supervised data to improve direct speech-to-speech translation. 1721-1725 - Linh The Nguyen, Nguyen Luong Tran, Long Doan, Manh Luong, Dat Quoc Nguyen:
A High-Quality and Large-Scale Dataset for English-Vietnamese Speech Translation. 1726-1730 - Qian Wang, Chen Wang, Jiajun Zhang:
Investigating Parameter Sharing in Multilingual Speech Translation. 1731-1735 - Zehui Yang, Yifan Chen, Lei Luo, Runyan Yang, Lingxuan Ye, Gaofeng Cheng, Ji Xu, Yaohui Jin, Qingqing Zhang, Pengyuan Zhang, Lei Xie, Yonghong Yan:
Open Source MagicData-RAMC: A Rich Annotated Mandarin Conversational(RAMC) Speech Dataset. 1736-1740 - Chengfei Li, Shuhao Deng, Yaoping Wang, Guangjing Wang, Yaguang Gong, Changbin Chen, Jinfeng Bai:
TALCS: An open-source Mandarin-English code-switching corpus and a speech recognition baseline. 1741-1745 - Keqi Deng, Shinji Watanabe, Jiatong Shi, Siddhant Arora:
Blockwise Streaming Transformer for Spoken Language Understanding and Simultaneous Speech Translation. 1746-1750 - Nguyen Luong Tran, Duong Minh Le, Dat Quoc Nguyen:
BARTpho: Pre-trained Sequence-to-Sequence Models for Vietnamese. 1751-1755 - Maxim Markitantov, Elena Ryumina, Dmitry Ryumin, Alexey Karpov:
Biometric Russian Audio-Visual Extended MASKS (BRAVE-MASKS) Corpus: Multimodal Mask Type Recognition Task. 1756-1760 - Jen-Tzung Chien, Yu-Han Huang:
Bayesian Transformer Using Disentangled Mask Attention. 1761-1765 - Hang Chen, Jun Du, Yusheng Dai, Chin-Hui Lee, Sabato Marco Siniscalchi, Shinji Watanabe, Odette Scharenborg, Jingdong Chen, Baocai Yin, Jia Pan:
Audio-Visual Speech Recognition in MISP2021 Challenge: Dataset Release and Deep Analysis. 1766-1770 - Danni Liu, Changhan Wang, Hongyu Gong, Xutai Ma, Yun Tang, Juan Miguel Pino:
From Start to Finish: Latency Reduction Strategies for Incremental Speech Synthesis in Simultaneous Speech-to-Speech Translation. 1771-1775 - Derek Tam, Surafel Melaku Lakew, Yogesh Virkar, Prashant Mathur, Marcello Federico:
Isochrony-Aware Neural Machine Translation for Automatic Dubbing. 1776-1780 - Qianqian Dong, Fengpeng Yue, Tom Ko, Mingxuan Wang, Qibing Bai, Yu Zhang:
Leveraging Pseudo-labeled Data to Improve Direct Speech-to-Speech Translation. 1781-1785
Source Separation I
- Zexu Pan, Meng Ge, Haizhou Li:
A Hybrid Continuity Loss to Reduce Over-Suppression for Time-domain Target Speaker Extraction. 1786-1790 - Axel Berg, Mark O'Connor, Kalle Åström, Magnus Oskarsson:
Extending GCC-PHAT using Shift Equivariant Neural Networks. 1791-1795 - Efthymios Tzinis, Gordon Wichern, Aswin Shanmugam Subramanian, Paris Smaragdis, Jonathan Le Roux:
Heterogeneous Target Speech Separation. 1796-1800 - Xubo Liu, Haohe Liu, Qiuqiang Kong, Xinhao Mei, Jinzheng Zhao, Qiushi Huang, Mark D. Plumbley, Wenwu Wang:
Separate What You Describe: Language-Queried Audio Source Separation. 1801-1805 - Dejan Markovic, Alexandre Défossez, Alexander Richard:
Implicit Neural Spatial Filtering for Multichannel Source Separation in the Waveform Domain. 1806-1810
ASR Technologies and Systems
- Jumon Nozaki, Tatsuya Kawahara, Kenkichi Ishizuka, Taiichi Hashimoto:
End-to-end Speech-to-Punctuated-Text Recognition. 1811-1815 - Adrien Pupier, Maximin Coavoux, Benjamin Lecouteux, Jérôme Goulian:
End-to-End Dependency Parsing of Spoken French. 1816-1820 - Shuo-Yiin Chang, Bo Li, Tara N. Sainath, Chao Zhang, Trevor Strohman, Qiao Liang, Yanzhang He:
Turn-Taking Prediction for Natural Conversational Speech. 1821-1825 - Shuo-Yiin Chang, Guru Prakash, Zelin Wu, Tara N. Sainath, Bo Li, Qiao Liang, Adam Stambler, Shyam Upadhyay, Manaal Faruqui, Trevor Strohman:
Streaming Intended Query Detection using E2E Modeling for Continued Conversation. 1826-1830 - Jan Lehecka, Jan Svec, Ales Prazák, Josef Psutka:
Exploring Capabilities of Monolingual Audio Transformers using Large Datasets in Automatic Speech Recognition of Czech. 1831-1835 - Rodrigo Schoburg Carrillo de Mira, Alexandros Haliassos, Stavros Petridis, Björn W. Schuller, Maja Pantic:
SVTS: Scalable Video-to-Speech Synthesis. 1836-1840
Speech Perception
- Takeshi Kishiyama, Chuyu Huang, Yuki Hirose:
One-step models in pitch perception: Experimental evidence from Japanese. 1841-1845 - Rubén Pérez Ramón, Martin Cooke, María Luisa García Lecumberri:
Generating iso-accented stimuli for second language research: methodology and a dataset for Spanish-accented English. 1846-1850 - Adrian Leemann, Péter Jeszenszky, Carina Steiner, Corinne Lanthemann:
Factors affecting the percept of Yanny v. Laurel (or mixed): Insights from a large-scale study on Swiss German listeners. 1851-1855 - Zhaoyan Zhang, Jason Zhang, Jody Kreiman:
Effects of laryngeal manipulations on voice gender perception. 1856-1860 - Boram Lee, Naomi Yamaguchi, Cécile Fougeron:
Why is Korean lenis stop difficult to perceive for L2 Korean learners? 1861-1865 - Alvaro Martin Iturralde Zurita, Meghan Clayards:
Lexical stress in Spanish word segmentation. 1866-1870
Spoken Term Detection and Voice Search
- Hyeon-Kyeong Shin, Hyewon Han, Doyeon Kim, Soo-Whan Chung, Hong-Goo Kang:
Learning Audio-Text Agreement for Open-vocabulary Keyword Spotting. 1871-1875 - Badr M. Abdullah, Bernd Möbius, Dietrich Klakow:
Integrating Form and Meaning: A Multi-Task Learning Model for Acoustic Word Embeddings. 1876-1880 - Seunghan Yang, Byeonggeun Kim, Inseop Chung, Simyung Chang:
Personalized Keyword Spotting through Multi-task Learning. 1881-1885 - Jan Svec, Jan Lehecka, Lubos Smídl:
Deep LSTM Spoken Term Detection using Wav2Vec 2.0 Recognizer. 1886-1890 - Christin Jose, Joe Wang, Grant P. Strimel, Mohammad Omar Khursheed, Yuriy Mishchenko, Brian Kulis:
Latency Control for Keyword Spotting. 1891-1895 - Prateeth Nayak, Takuya Higuchi, Anmol Gupta, Shivesh Ranjan, Stephen Shum, Siddharth Sigtia, Erik Marchi, Varun Lakshminarasimhan, Minsik Cho, Saurabh Adya, Chandra Dhir, Ahmed H. Tewfik:
Improving Voice Trigger Detection with Metric Learning. 1896-1900
Speech and Language in Health: From Remote Monitoring to Medical Conversations I
- Hagen Soltau, Izhak Shafran, Mingqiu Wang, Laurent El Shafey:
RNN Transducers for Named Entity Recognition with constraints on alignment for understanding medical conversations. 1901-1905 - Zixiu Wu, Rim Helaoui, Diego Reforgiato Recupero, Daniele Riboni:
Towards Automated Counselling Decision-Making: Remarks on Therapist Action Forecasting on the AnnoMI Dataset. 1906-1910 - Salvatore Fara, Stefano Goria, Emilia Molimpakis, Nicholas Cummins:
Speech and the n-Back task as a lens into depression. How combining both may allow us to isolate different core symptoms of depression. 1911-1915 - Amrit Romana, Minxue Niu, Matthew Perez, Angela Roberts, Emily Mower Provost:
Enabling Off-the-Shelf Disfluency Detection and Categorization for Pathological Speech. 1916-1920 - Catarina Botelho, Tanja Schultz, Alberto Abad, Isabel Trancoso:
Challenges of using longitudinal and cross-domain corpora on studies of pathological speech. 1921-1925
Speech Synthesis: Linguistic Processing, Paradigms and Other Topics I
- Yi-Chang Chen, Yu-Chuan Steven, Yen-Cheng Chang, Yi-Ren Yeh:
g2pW: A Conditional Weighted Softmax BERT for Polyphone Disambiguation in Mandarin. 1926-1930 - Byeongseon Park, Ryuichi Yamamoto, Kentaro Tachibana:
A Unified Accent Estimation Method Based on Multi-Task Learning for Japanese Text-to-Speech. 1931-1935 - Tuomo Raitio, Petko Petkov, Jiangchuan Li, P. V. Muhammed Shifas, Andrea Davis, Yannis Stylianou:
Vocal effort modeling in neural TTS for improving the intelligibility of synthetic speech in noise. 1936-1940 - Eunwoo Song, Ryuichi Yamamoto, Ohsung Kwon, Chan-Ho Song, Min-Jae Hwang, Suhyeon Oh, Hyun-Wook Yoon, Jin-Seob Kim, Jae-Min Kim:
TTS-by-TTS 2: Data-Selective Augmentation for Neural Speech Synthesis Using Ranking Support Vector Machine with Variational Autoencoder. 1941-1945 - Giulia Comini, Goeric Huybrechts, Manuel Sam Ribeiro, Adam Gabrys, Jaime Lorenzo-Trueba:
Low-data? No problem: low-resource, language-agnostic conversational text-to-speech via F0-conditioned data augmentation. 1946-1950
Show and Tell II
- Digvijay Ingle, Ayush Kumar, Krishnachaitanya Gogineni, Jithendra Vepa:
Real-Time Monitoring of Silences in Contact Center Conversations. 1951-1952 - Konrad Zielinski, Marek Grzelec, Martin Hagmüller:
Humanizing bionic voice: interactive demonstration of aesthetic design and control factors influencing the devices assembly and waveshape engineering. 1953-1954 - Damien Ronssin, Milos Cernak:
Application for Real-time Personalized Speaker Extraction. 1955-1956 - Debarpan Bhattacharya, Debottam Dutta, Neeraj Kumar Sharma, Srikanth Raj Chetupalli, Pravin Mote, Sriram Ganapathy, Chandrakiran C, Sahiti Nori, Suhail K. K, Sadhana Gonuguntla, Murali Alagesan:
Coswara: A website application enabling COVID-19 screening by analysing respiratory sound samples and health symptoms. 1957-1958 - P. Schäfer, Paula Andrea Pérez-Toro, Philipp Klumpp, Juan Rafael Orozco-Arroyave, Elmar Nöth, Andreas K. Maier, A. Abad, Maria Schuster, Tomás Arias-Vergara:
CoachLea: an Android Application to Evaluate the Speech Production and Perception of Children with Hearing Loss. 1959-1960 - Fasih Haider, Saturnino Luz:
An Automated Mood Diary for Older User's using Ambient Assisted Living Recorded Speech. 1961-1962
Multimodal Speech Emotion Recognition and Paralinguistics
- Hai-tao Xu, Jie Zhang, Li-Rong Dai:
Differential Time-frequency Log-mel Spectrogram Features for Vision Transformer Based Infant Cry Recognition. 1963-1967 - Daniel Fernau, Stefan Hillmann, Nils Feldhus, Tim Polzehl:
Towards Automated Dialog Personalization using MBTI Personality Indicators. 1968-1972 - Fan Qian, Hongwei Song, Jiqing Han:
Word-wise Sparse Attention for Multimodal Sentiment Analysis. 1973-1977 - Tarun Gupta, Duc-Tuan Truong, Tran The Anh, Eng Siong Chng:
Estimation of speaker age and height from speech signal using bi-encoder transformer mixture model. 1978-1982 - Weiqiao Zheng, Ping Yang, Rongfeng Lai, Kongyang Zhu, Tao Zhang, Junpeng Zhang, Hongcheng Fu:
Exploring Multi-task Learning Based Gender Recognition and Age Estimation for Class-imbalanced Data. 1983-1987 - Jie Wei, Guanyu Hu, Xinyu Yang, Anh Tuan Luu, Yizhuo Dong:
Audio-Visual Domain Adaptation Feature Fusion for Speech Emotion Recognition. 1988-1992 - Minyue Zhang, Hongwei Ding:
Impact of Background Noise and Contribution of Visual Information in Emotion Identification by Native Mandarin Speakers. 1993-1997 - Wei Yang, Satoru Fukayama, Panikos Heracleous, Jun Ogata:
Exploiting Fine-tuning of Self-supervised Learning Models for Improving Bi-modal Sentiment Analysis and Emotion Recognition. 1998-2002 - Dehua Tao, Tan Lee, Harold Chui, Sarah Luk:
Characterizing Therapist's Speaking Style in Relation to Empathy in Psychotherapy. 2003-2007 - Dehua Tao, Tan Lee, Harold Chui, Sarah Luk:
Hierarchical Attention Network for Evaluating Therapist Empathy in Counseling Session. 2008-2012 - Jinchao Li, Shuai Wang, Yang Chao, Xunying Liu, Helen Meng:
Context-aware Multimodal Fusion for Emotion Recognition. 2013-2017 - Jinhan Wang, Vijay Ravi, Jonathan Flint, Abeer Alwan:
Unsupervised Instance Discriminative Learning for Depression Detection from Speech Signals. 2018-2022 - Nasim Mahdinazhad Sardhaei, Marzena Zygis, Hamid Sharifzadeh:
How do our eyebrows respond to masks and whispering? The case of Persians. 2023-2027 - Alice Baird, Panagiotis Tzirakis, Jeffrey A. Brooks, Lauren Kim, Michael Opara, Christopher B. Gregory, Jacob Metrick, Garrett Boseck, Dacher Keltner, Alan Cowen:
State & Trait Measurement from Nonverbal Vocalizations: A Multi-Task Joint Learning Approach. 2028-2032 - Amruta Saraf, Ganesh Sivaraman, Elie Khoury:
Confidence Measure for Automatic Age Estimation From Speech. 2033-2037
Neural Transducers, Streaming ASR and Novel ASR Models
- Andrea Fasoli, Chia-Yu Chen, Mauricio J. Serrano, Swagath Venkataramani, George Saon, Xiaodong Cui, Brian Kingsbury, Kailash Gopalakrishnan:
Accelerating Inference and Language Model Fusion of Recurrent Neural Network Transducers via End-to-End 4-bit Quantization. 2038-2042 - Guangzhi Sun, Chao Zhang, Philip C. Woodland:
Tree-constrained Pointer Generator with Graph Neural Network Encodings for Contextual Speech Recognition. 2043-2047 - Junfeng Hou, Jinkun Chen, Wanyu Li, Yufeng Tang, Jun Zhang, Zejun Ma:
Bring dialogue-context into RNN-T for streaming ASR. 2048-2052 - Felix Weninger, Marco Gaudesi, Md. Akmal Haidar, Nicola Ferri, Jesús Andrés-Ferrer, Puming Zhan:
Conformer with dual-mode chunked attention for joint online and offline ASR. 2053-2057 - Wei Zhou, Wilfried Michel, Ralf Schlüter, Hermann Ney:
Efficient Training of Neural Transducer for Speech Recognition. 2058-2062 - Zhifu Gao, Shiliang Zhang, Ian McLoughlin, Zhijie Yan:
Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition. 2063-2067 - Fangjun Kuang, Liyong Guo, Wei Kang, Long Lin, Mingshuang Luo, Zengwei Yao, Daniel Povey:
Pruned RNN-T for fast, memory-efficient ASR training. 2068-2072 - Xianchao Wu:
Deep Sparse Conformer for Speech Recognition. 2073-2077 - Hung-Shin Lee, Pin-Tuan Huang, Yao-Fei Cheng, Hsin-Min Wang:
Chain-based Discriminative Autoencoders for Speech Recognition. 2078-2082 - Jay Mahadeokar, Yangyang Shi, Ke Li, Duc Le, Jiedan Zhu, Vikas Chandra, Ozlem Kalinli, Michael L. Seltzer:
Streaming parallel transducer beam search with fast slow cascaded encoders. 2083-2087 - Mohan Li, Rama Sanand Doddipatla, Catalin Zorila:
Self-regularised Minimum Latency Training for Streaming Transformer-based Speech Recognition. 2088-2092 - Dario Albesano, Jesús Andrés-Ferrer, Nicola Ferri, Puming Zhan:
On the Prediction Network Architecture in RNN-T for ASR. 2093-2097 - Yusuke Shinohara, Shinji Watanabe:
Minimum latency training of sequence transducers for streaming end-to-end speech recognition. 2098-2102 - Keyu An, Huahuan Zheng, Zhijian Ou, Hongyu Xiang, Ke Ding, Guanglu Wan:
CUSIDE: Chunking, Simulating Future Context and Decoding for Streaming ASR. 2103-2107 - Xianchao Wu:
Attention Enhanced Citrinet for Speech Recognition. 2108-2112
Zero, Low-resource and Multi-Modal Speech Recognition II
- Qiantong Xu, Alexei Baevski, Michael Auli:
Simple and Effective Zero-shot Cross-lingual Phoneme Recognition. 2113-2117 - Bowen Shi, Wei-Ning Hsu, Abdelrahman Mohamed:
Robust Self-Supervised Audio-Visual Speech Recognition. 2118-2122 - Robin Algayres, Adel Nabli, Benoît Sagot, Emmanuel Dupoux:
Speech Sequence Embeddings using Nearest Neighbors Contrastive Learning. 2123-2127 - Junhao Xu, Shoukang Hu, Xunying Liu, Helen Meng:
Towards Green ASR: Lossless 4-bit Quantization of a Hybrid TDNN System on the 300-hr Swithboard Corpus. 2128-2132 - Siqing Qin, Longbiao Wang, Sheng Li, Yuqin Lin, Jianwu Dang:
Finer-grained Modeling units-based Meta-Learning for Low-resource Tibetan Speech Recognition. 2133-2137
Atypical Speech Analysis and Detection
- Parvaneh Janbakhshi, Ina Kodrasi:
Adversarial-Free Speaker Identity-Invariant Representation Learning for Automatic Dysarthric Speech Classification. 2138-2142 - Zhenglin Zhang, Lizhuang Yang, Xun Wang, Hai Li:
Automated Detection of Wilson's Disease Based on Improved Mel-frequency Cepstral Coefficients with Signal Decomposition. 2143-2147 - Zixia Fan, Jing Shao, Weigong Pan, Min Xu, Lan Wang:
The effect of backward noise on lexical tone discrimination in Mandarin-speaking amusics. 2148-2152 - Xiaoquan Ke, Man-Wai Mak, Helen M. Meng:
Automatic Selection of Discriminative Features for Dementia Detection in Cantonese-Speaking People. 2153-2157 - Zhuoya Liu, Mark A. Huckvale, Julian McGlashan:
Automated Voice Pathology Discrimination from Continuous Speech Benefits from Analysis by Phonetic Context. 2158-2162 - Adria Mallol-Ragolta, Helena Cuesta, Emilia Gómez, Björn W. Schuller:
Multi-Type Outer Product-Based Fusion of Respiratory Sounds for Detecting COVID-19. 2163-2167 - Xueshuai Zhang, Jiakun Shen, Jun Zhou, Pengyuan Zhang, Yonghong Yan, Zhihua Huang, Yanfen Tang, Yu Wang, Fujie Zhang, Shaoxing Zhang, Aijun Sun:
Robust Cough Feature Extraction and Classification Method for COVID-19 Cough Detection Based on Vocalization Characteristics. 2168-2172 - Farhad Javanmardi, Sudarsana Reddy Kadiri, Manila Kodali, Paavo Alku:
Comparing 1-dimensional and 2-dimensional spectral feature representations in voice pathology detection using machine learning and deep learning classifiers. 2173-2177 - Gerasimos Chatzoudis, Manos Plitsis, Spyridoula Stamouli, Athanasia-Lida Dimou, Nassos Katsamanis, Vassilis Katsouros:
Zero-Shot Cross-lingual Aphasia Detection using Automatic Speech Recognition. 2178-2182 - Youxiang Zhu, Xiaohui Liang, John A. Batsis, Robert M. Roth:
Domain-aware Intermediate Pretraining for Dementia Detection with Limited Data. 2183-2187 - Cécile Fougeron, Nicolas Audibert, Ina Kodrasi, Parvaneh Janbakhshi, Michaela Pernon, Nathalie Lévêque, Stephanie Borel, Marina Laganaro, Hervé Bourlard, Frédéric Assal:
Comparison of 5 methods for the evaluation of intelligibility in mild to moderate French dysarthric speech. 2188-2192
Adaptation, Transfer Learning, and Distillation for ASR
- Kuan-Po Huang, Yu-Kuan Fu, Yu Zhang, Hung-yi Lee:
Improving Distortion Robustness of Self-supervised Speech Processing Tasks with Domain Adaptation. 2193-2197 - Guan-Ting Lin, Shang-Wen Li, Hung-yi Lee:
Listen, Adapt, Better WER: Source-free Single-utterance Test-time Adaptation for Automatic Speech Recognition. 2198-2202 - Kwanghee Choi, Hyung-Min Park:
Distilling a Pretrained Language Model to a Multilingual ASR Model. 2203-2207 - Hiroaki Sato, Tomoyasu Komori, Takeshi Mishima, Yoshihiko Kawai, Takahiro Mochizuki, Shoei Sato, Tetsuji Ogawa:
Text-Only Domain Adaptation Based on Intermediate CTC. 2208-2212 - Jenthe Thienpondt, Kris Demuynck:
Transfer Learning for Robust Low-Resource Children's Speech ASR with Transformers and Source-Filter Warping. 2213-2217 - Yuki Takashima, Shota Horiguchi, Shinji Watanabe, Leibny Paola García-Perera, Yohei Kawaguchi:
Updating Only Encoders Prevents Catastrophic Forgetting of End-to-End ASR Models. 2218-2222
Speaker and Language Recognition I
- Jeong-Hwan Choi, Joon-Young Yang, Ye-Rin Jeoung, Joon-Hyuk Chang:
Improved CNN-Transformer using Broadcasted Residual Learning for Text-Independent Speaker Verification. 2223-2227 - Jee-weon Jung, You Jin Kim, Hee-Soo Heo, Bong-Jin Lee, Youngki Kwon, Joon Son Chung:
Pushing the limits of raw waveform speaker recognition. 2228-2232 - Hexin Liu, Leibny Paola García-Perera, Andy W. H. Khong, Suzy J. Styles, Sanjeev Khudanpur:
PHO-LID: A Unified Model Incorporating Acoustic-Phonetic and Phonotactic Information for Language Identification. 2233-2237 - Moakala Tzudir, Priyankoo Sarmah, S. R. Mahadeva Prasanna:
Prosodic Information in Dialect Identification of a Tonal Language: The case of Ao. 2238-2242 - Wo Jae Lee, Emanuele Coviello:
A Multimodal Strategy for Singing Language Identification. 2243-2247
Pathological Speech Analysis
- Khalid Daoudi, Biswajit Das, Solange Milhé de Saint Victor, Alexandra Foubert-Samier, Margherita Fabbri, Anne Pavy-Le Traon, Olivier Rascol, Virginie Woisard, Wassilios G. Meissner:
A comparative study on vowel articulation in Parkinson's disease and multiple system atrophy. 2248-2252 - Luc Ardaillon, Nathalie Henrich Bernardoni, Olivier Perrotin:
Voicing decision based on phonemes classification and spectral moments for whisper-to-speech conversion. 2253-2257 - Tanya Talkar, Christina Manxhari, James J. Williamson, Kara M. Smith, Thomas F. Quatieri:
Speech Acoustics in Mild Cognitive Impairment and Parkinson's Disease With and Without Concurrent Drawing Tasks. 2258-2262 - Kelvin Tran, Lingfeng Xu, Gabriela Stegmann, Julie Liss, Visar Berisha, Rene Utianski:
Investigating the Impact of Speech Compression on the Acoustics of Dysarthric Speech. 2263-2267 - Avamarie Brueggeman, John H. L. Hansen:
Speaker Trait Enhancement for Cochlear Implant Users: A Case Study for Speaker Emotion Perception. 2268-2272 - Neha Reddy, Yoonjeong Lee, Zhaoyan Zhang, Dinesh K. Chhetri:
Optimal thyroplasty implant shape and stiffness for treatment of acute unilateral vocal fold paralysis: Evidence from a canine in vivo phonation model. 2273-2277
Cross/Multi-lingual ASR
- Arun Babu, Changhan Wang, Andros Tjandra, Kushal Lakhotia, Qiantong Xu, Naman Goyal, Kritika Singh, Patrick von Platen, Yatharth Saraf, Juan Pino, Alexei Baevski, Alexis Conneau, Michael Auli:
XLS-R: Self-supervised Cross-lingual Speech Representation Learning at Scale. 2278-2282 - Janine Rugayan, Torbjørn Svendsen, Giampiero Salvi:
Semantically Meaningful Metrics for Norwegian ASR Systems. 2283-2287 - Ondrej Klejch, Electra Wallington, Peter Bell:
Deciphering Speech: a Zero-Resource Approach to Cross-Lingual Transfer in ASR. 2288-2292 - Rishabh Kumar, Devaraja Adiga, Rishav Ranjan, Amrith Krishna, Ganesh Ramakrishnan, Pawan Goyal, Preethi Jyothi:
Linguistically Informed Post-processing for ASR Error correction in Sanskrit. 2293-2297 - Mahir Morshed, Mark Hasegawa-Johnson:
Cross-lingual articulatory feature information transfer for speech recognition using recurrent progressive neural networks. 2298-2302
Speaking Styles and Interaction Styles I
- Diego Aguirre, Nigel G. Ward, Jonathan E. Avila, Heike Lehnert-LeHouillier:
Comparison of Models for Detecting Off-Putting Speaking Styles. 2303-2307 - Seiya Kawano, Muteki Arioka, Akishige Yuguchi, Kenta Yamamoto, Koji Inoue, Tatsuya Kawahara, Satoshi Nakamura, Koichiro Yoshino:
Multimodal Persuasive Dialogue Corpus using Teleoperated Android. 2308-2312 - Yookyung Shin, Younggun Lee, Suhee Jo, Yeongtae Hwang, Taesu Kim:
Text-driven Emotional Style Control and Cross-speaker Style Transfer in Neural TTS. 2313-2317 - Adaeze O. Adigwe, Esther Klabbers:
Strategies for developing a Conversational Speech Dataset for Text-To-Speech Synthesis. 2318-2322 - Xiyuan Gao, Shekhar Nayak, Matt Coler:
Deep CNN-based Inductive Transfer Learning for Sarcasm Detection in Speech. 2323-2327
Speaking Styles and Interaction Styles II
- Kentaro Mitsui, Tianyu Zhao, Kei Sawada, Yukiya Hono, Yoshihiko Nankaku, Keiichi Tokuda:
End-to-End Text-to-Speech Based on Latent Representation of Speaking Styles Using Spontaneous Dialogue. 2328-2332 - Amber Afshan, Abeer Alwan:
Attention-based conditioning methods using variable frame rate for style-robust speaker verification. 2333-2337 - Amber Afshan, Abeer Alwan:
Learning from human perception to improve automatic speaker verification in style-mismatched conditions. 2338-2342 - Katariina Martikainen, Jussi Karlgren, Khiet Truong:
Exploring audio-based stylistic variation in podcasts. 2343-2347
Speech Synthesis: Tools, Data, and Evaluation
- Kamil Deja, Ariadna Sánchez, Julian Roth, Marius Cotescu:
Automatic Evaluation of Speaker Similarity. 2348-2352 - Ziyao Zhang, Alessio Falai, Ariadna Sánchez, Orazio Angelini, Kayoko Yanagisawa:
Mix and Match: An Empirical Study on Training Corpus Composition for Polyglot Text-To-Speech (TTS). 2353-2357 - Shinnosuke Takamichi, Wataru Nakata, Naoko Tanji, Hiroshi Saruwatari:
J-MAC: Japanese multi-speaker audiobook corpus for speech synthesis. 2358-2362 - Jacob Webber, Samuel K. Lo, Isaac L. Bleaman:
REYD - The First Yiddish Text-to-Speech Dataset and System. 2363-2367 - Marcel de Korte, Jaebok Kim, Aki Kunikoshi, Adaeze Adigwe, Esther Klabbers:
Data-augmented cross-lingual synthesis in a teacher-student framework. 2368-2372 - Ayushi Pandey, Sébastien Le Maguer, Julie Carson-Berndsen, Naomi Harte:
Production characteristics of obstruents in WaveNet and older TTS systems. 2373-2377 - Sébastien Le Maguer, Simon King, Naomi Harte:
Back to the Future: Extending the Blizzard Challenge 2013. 2378-2382 - Josh Meyer, David Ifeoluwa Adelani, Edresson Casanova, Alp Öktem, Daniel Whitenack, Julian Weber, Salomon Kabongo, Elizabeth Salesky, Iroro Orife, Colin Leong, Perez Ogayo, Chris Chinenye Emezue, Jonathan Mukiibi, Salomey Osei, Apelete Agbolo, Victor Akinode, Bernard Opoku, Samuel Olanrewaju, Jesujoba O. Alabi, Shamsuddeen Hassan Muhammad:
BibleTTS: a large, high-fidelity, multilingual, and uniquely African speech corpus. 2383-2387 - Georgia Maniati, Alexandra Vioni, Nikolaos Ellinas, Karolos Nikitaras, Konstantinos Klapsas, June Sig Sung, Gunu Jho, Aimilios Chalamandaris, Pirros Tsiakoulis:
SOMOS: The Samsung Open MOS Dataset for the Evaluation of Neural Text-to-Speech Synthesis. 2388-2392
Acoustic Signal Representation and Analysis II
- Byeonggeun Kim, Seunghan Yang, Jangho Kim, Hyunsin Park, Juntae Lee, Simyung Chang:
Domain Generalization with Relaxed Instance Frequency-wise Normalization for Multi-device Acoustic Scene Classification. 2393-2397 - Rui Tao, Long Yan, Kazushige Ouchi, Xiangdong Wang:
Couple learning for semi-supervised sound event detection. 2398-2402 - Rajeev Rajan, Ananya Ayasi:
Oktoechos Classification in Liturgical Music Using SBU-LSTM/GRU. 2403-2407 - Yuhang He, Andrew Markham:
SoundDoA: Learn Sound Source Direction of Arrival and Semantics from Sound Raw Waveforms. 2408-2412 - Christian Bergler, Alexander Barnhill, Dominik Perrin, Manuel Schmitt, Andreas K. Maier, Elmar Nöth:
ORCA-WHISPER: An Automatic Killer Whale Sound Type Generation Toolkit Using Deep Learning. 2413-2417 - Joon-Hyuk Chang, Won-Gook Choi:
Convolutional Recurrent Neural Network with Auxiliary Stream for Robust Variable-Length Acoustic Scene Classification. 2418-2422 - Shahaf Bassan, Yossi Adi, Jeffrey S. Rosenschein:
Unsupervised Symbolic Music Segmentation using Ensemble Temporal Prediction Errors. 2423-2427 - Amir Shirian, Krishna Somandepalli, Victor Sanchez, Tanaya Guha:
Visually-aware Acoustic Event Detection using Heterogeneous Graphs. 2428-2432 - Arshdeep Singh, Mark D. Plumbley:
A Passive Similarity based CNN Filter Pruning for Efficient Acoustic Scene Classification. 2433-2437 - Alan Baade, Puyuan Peng, David Harwath:
MAE-AST: Masked Autoencoding Audio Spectrogram Transformer. 2438-2442
Speech and Language in Health: From Remote Monitoring to Medical Conversations II
- Sebastian Peter Bayerl, Gabriel Roccabruna, Shammur Absar Chowdhury, Tommaso Ciulli, Morena Danieli, Korbinian Riedhammer, Giuseppe Riccardi:
What can Speech and Language Tell us About the Working Alliance in Psychotherapy. 2443-2447 - Geoffrey T. Frost, Grant Theron, Thomas Niesler:
TB or not TB? Acoustic cough analysis for tuberculosis classification. 2448-2452 - Visar Berisha, Chelsea Krantsevich, Gabriela Stegmann, Shira Hahn, Julie Liss:
Are reported accuracies in the clinical speech machine learning literature overoptimistic? 2453-2457 - Bahman Mirheidari, André Bittar, Nicholas Cummins, Johnny Downs, Helen L. Fisher, Heidi Christensen:
Automatic Detection of Expressed Emotion from Five-Minute Speech Samples: Challenges and Opportunities. 2458-2462 - Bahman Mirheidari, Daniel Blackburn, Heidi Christensen:
Automatic cognitive assessment: Combining sparse datasets with disparate cognitive scores. 2463-2467 - Ting Dang, Thomas Quinnell, Cecilia Mascolo:
Exploring Semi-supervised Learning for Audio-based COVID-19 Detection using FixMatch. 2468-2472 - Debarpan Bhattacharya, Debottam Dutta, Neeraj Kumar Sharma, Srikanth Raj Chetupalli, Pravin Mote, Sriram Ganapathy, Chandrakiran C, Sahiti Nori, Suhail K. K, Sadhana Gonuguntla, Murali Alagesan:
Analyzing the impact of SARS-CoV-2 variants on respiratory sound signals. 2473-2477 - Franziska Braun, Markus Förstel, Bastian Oppermann, Andreas Erzigkeit, Hartmut Lehfeld, Thomas Hillemacher, Korbinian Riedhammer:
Automated Evaluation of Standardized Dementia Screening Tests. 2478-2482 - Paula Andrea Pérez-Toro, Philipp Klumpp, Abner Hernandez, Tomas Arias, Patricia Lillo, Andrea Slachevsky, Adolfo Martín García, Maria Schuster, Andreas K. Maier, Elmar Nöth, Juan Rafael Orozco-Arroyave:
Alzheimer's Detection from English to Spanish Using Acoustic and Linguistic Embeddings. 2483-2487 - Jing Su, Longxiang Zhang, Hamid Reza Hassanzadeh, Thomas Schaaf:
Extract and Abstract with BART for Clinical Notes from Doctor-Patient Conversations. 2488-2492 - Bishal Lamichhane, Nidal Moukaddam, Ankit B. Patel, Ashutosh Sabharwal:
Dyadic Interaction Assessment from Free-living Audio for Depression Severity Assessment. 2493-2497 - Venkata Srikanth Nallanthighal, Aki Härmä, Helmer Strik:
COVID-19 detection based on respiratory sensing from speech. 2498-2502
Dereverberation and Echo Cancellation
- Xiaoxue Luo, Chengshi Zheng, Andong Li, Yuxuan Ke, Xiaodong Li:
Bifurcation and Reunion: A Loss-Guided Two-Stage Approach for Monaural Speech Dereverberation. 2503-2507 - Linjuan Cheng, Chengshi Zheng, Andong Li, Yuquan Wu, Renhua Peng, Xiaodong Li:
A deep complex multi-frame filtering network for stereophonic acoustic echo cancellation. 2508-2512 - Chang Han, Weiping Tu, Yuhong Yang, Jingyi Li, Xinhong Li:
Speaker- and Phone-aware Convolutional Transformer Network for Acoustic Echo Cancellation. 2513-2517 - Shimin Zhang, Ziteng Wang, Yukai Ju, Yihui Fu, Yueyue Na, Qiang Fu, Lei Xie:
Personalized Acoustic Echo Cancellation for Full-duplex Communications. 2518-2522 - Chenggang Zhang, Jinjiang Liu, Xueliang Zhang:
LCSM: A Lightweight Complex Spectral Mapping Framework for Stereophonic Acoustic Echo Cancellation. 2523-2527 - Vinay Kothapally, Yong Xu, Meng Yu, Shi-Xiong Zhang, Dong Yu:
Joint Neural AEC and Beamforming with Double-Talk Detection. 2528-2532 - Karim Helwani, Erfan Soltanmohammadi, Michael Mark Goodwin, Arvindh Krishnaswamy:
Clock Skew Robust Acoustic Echo Cancellation. 2533-2537 - Sankaran Panchapagesan, Arun Narayanan, Turaj Zakizadeh Shabestary, Shuai Shao, Nathan Howard, Alex Park, James Walker, Alexander Gruenstein:
A Conformer-based Waveform-domain Neural Acoustic Echo Canceller Optimized for ASR Accuracy. 2538-2542 - Vinay Kothapally, John H. L. Hansen:
Complex-Valued Time-Frequency Self-Attention for Speech Dereverberation. 2543-2547
Voice Conversion and Adaptation III
- Liumeng Xue, Shan Yang, Na Hu, Dan Su, Lei Xie:
Learning Noise-independent Speech Representation for High-quality Voice Conversion for Noisy Target Speakers. 2548-2552 - Sicheng Yang, Methawee Tantrawenith, Haolin Zhuang, Zhiyong Wu, Aolan Sun, Jianzong Wang, Ning Cheng, Huaizhen Tang, Xintao Zhao, Jie Wang, Helen Meng:
Speech Representation Disentanglement with Adversarial Mutual Information Learning for One-shot Voice Conversion. 2553-2557 - Jiahong Huang, Wen Xu, Yule Li, Junshi Liu, Dongpeng Ma, Wei Xiang:
FlowCPCVC: A Contrastive Predictive Coding Supervised Flow Framework for Any-to-Any Voice Conversion. 2558-2562 - Yi Lei, Shan Yang, Jian Cong, Lei Xie, Dan Su:
Glow-WaveGAN 2: High-quality Zero-shot Text-to-speech Synthesis and Any-to-any Voice Conversion. 2563-2567 - Yihan Wu, Xu Tan, Bohan Li, Lei He, Sheng Zhao, Ruihua Song, Tao Qin, Tie-Yan Liu:
AdaSpeech 4: Adaptive Text to Speech in Zero-Shot Scenarios. 2568-2572 - Yixuan Zhou, Changhe Song, Xiang Li, Luwen Zhang, Zhiyong Wu, Yanyao Bian, Dan Su, Helen Meng:
Content-Dependent Fine-Grained Speaker Embedding for Zero-Shot Speaker Adaptation in Text-to-Speech Synthesis. 2573-2577 - Haoquan Yang, Liqun Deng, Yu Ting Yeung, Nianzu Zheng, Yong Xu:
Streamable Speech Representation Disentanglement and Multi-Level Prosody Modeling for Live One-Shot Voice Conversion. 2578-2582 - Tuan-Nam Nguyen, Ngoc-Quan Pham, Alexander Waibel:
Accent Conversion using Pre-trained Model and Synthesized Data from Voice Conversion. 2583-2587 - Pol van Rijn, Silvan Mertes, Dominik Schiller, Piotr Dura, Hubert Siuzdak, Peter M. C. Harrison, Elisabeth André, Nori Jacoby:
VoiceMe: Personalized voice generation in TTS. 2588-2592 - Ruibin Yuan, Yuxuan Wu, Jacob Li, Jaxter Kim:
DeID-VC: Speaker De-identification via Zero-shot Pseudo Voice Conversion. 2593-2597 - Jiachen Lian, Chunlei Zhang, Gopala Krishna Anumanchipalli, Dong Yu:
Towards Improved Zero-shot Voice Conversion with Conditional DSVAE. 2598-2602 - Zongyang Du, Berrak Sisman, Kun Zhou, Haizhou Li:
Disentanglement of Emotional Style and Speaker Identity for Expressive Voice Conversion. 2603-2607
Novel Models and Training Methods for ASR III
- Zhong Meng, Yashesh Gaur, Naoyuki Kanda, Jinyu Li, Xie Chen, Yu Wu, Yifan Gong:
Internal Language Model Adaptation with Text-Only Data for End-to-End Speech Recognition. 2608-2612 - Ye-Qian Du, Jie Zhang, Qiu-Shi Zhu, Lirong Dai, Ming-Hui Wu, Xin Fang, Zhou-Wang Yang:
A Complementary Joint Training Approach Using Unpaired Speech and Text A Complementary Joint Training Approach Using Unpaired Speech and Text. 2613-2617 - Xun Gong, Zhikai Zhou, Yanmin Qian:
Knowledge Transfer and Distillation from Autoregressive to Non-Autoregessive Speech Recognition. 2618-2622 - Jiajun Deng, Xurong Xie, Tianzi Wang, Mingyu Cui, Boyang Xue, Zengrui Jin, Mengzhe Geng, Guinan Li, Xunying Liu, Helen Meng:
Confidence Score Based Conformer Speaker Adaptation for Speech Recognition. 2623-2627 - Han Zhu, Jindong Wang, Gaofeng Cheng, Pengyuan Zhang, Yonghong Yan:
Decoupled Federated Learning for ASR with Non-IID Data. 2628-2632 - Sanli Tian, Keqi Deng, Zehan Li, Lingxuan Ye, Gaofeng Cheng, Ta Li, Yonghong Yan:
Knowledge Distillation For CTC-based Speech Recognition Via Consistent Acoustic Representation Learning. 2633-2637 - Xiaodong Cui, George Saon, Tohru Nagano, Masayuki Suzuki, Takashi Fukuda, Brian Kingsbury, Gakuto Kurata:
Improving Generalization of Deep Neural Network Acoustic Models with Length Perturbation and N-best Based Label Smoothing. 2638-2642 - Chengyi Wang, Yiming Wang, Yu Wu, Sanyuan Chen, Jinyu Li, Shujie Liu, Furu Wei:
Supervision-Guided Codebooks for Masked Prediction in Speech Pre-training. 2643-2647 - Shuo Ren, Shujie Liu, Yu Wu, Long Zhou, Furu Wei:
Speech Pre-training with Acoustic Piece. 2648-2652 - Bowen Zhang, Songjun Cao, Xiaoming Zhang, Yike Zhang, Long Ma, Takahiro Shinozaki:
Censer: Curriculum Semi-supervised Learning for Speech Recognition Based on Self-supervised Pre-training. 2653-2657 - Junyi Ao, Ziqiang Zhang, Long Zhou, Shujie Liu, Haizhou Li, Tom Ko, Lirong Dai, Jinyu Li, Yao Qian, Furu Wei:
Pre-Training Transformer Decoder for End-to-End ASR Model with Unpaired Speech Data. 2658-2662 - Ramit Sawhney, Megh Thakkar, Vishwa Shah, Puneet Mathur, Vasu Sharma, Dinesh Manocha:
PISA: PoIncaré Saliency-Aware Interpolative Augmentation. 2663-2667 - Muqiao Yang, Ian R. Lane, Shinji Watanabe:
Online Continual Learning of End-to-End Speech Recognition Models. 2668-2672 - Takafumi Moriya, Hiroshi Sato, Tsubasa Ochiai, Marc Delcroix, Takahiro Shinozaki:
Streaming Target-Speaker ASR with Neural Transducer. 2673-2677 - Arjit Jain, Pranay Reddy Samala, Deepak Mittal, Preethi Jyothi, Maneesh Singh:
SPLICEOUT: A Simple and Efficient Audio Augmentation Method. 2678-2682
Spoken Language Modeling and Understanding
- Vishal Sunder, Eric Fosler-Lussier, Samuel Thomas, Hong-Kwang Kuo, Brian Kingsbury:
Tokenwise Contrastive Pretraining for Finer Speech-to-BERT Alignment in End-to-End Speech-to-Intent Systems. 2683-2687 - Yasuhito Ohsugi, Itsumi Saito, Kyosuke Nishida, Sen Yoshida:
Japanese ASR-Robust Pre-trained Language Model with Pseudo-Error Sentences Generated by Grapheme-Phoneme Conversion. 2688-2692 - Jingjing Dong, Jiayi Fu, Peng Zhou, Hao Li, Xiaorui Wang:
Improving Spoken Language Understanding with Cross-Modal Contrastive Learning. 2693-2697 - Anderson R. Avila, Khalil Bibi, Rui Heng Yang, Xinlin Li, Chao Xing, Xiao Chen:
Low-bit Shift Network for End-to-End Spoken Language Understanding. 2698-2702 - Yingying Gao, Junlan Feng, Chao Deng, Shilei Zhang:
Meta Auxiliary Learning for Low-resource Spoken Language Understanding. 2703-2707 - Ye Wang, Baishun Ling, Yanmeng Wang, Junhao Xue, Shaojun Wang, Jing Xiao:
Adversarial Knowledge Distillation For Robust Spoken Language Understanding. 2708-2712 - Yangyang Ou, Peng Zhang, Jing Zhang, Hui Gao, Xing Ma:
Incorporating Dual-Aware with Hierarchical Interactive Memory Networks for Task-Oriented Dialogue. 2713-2717 - Yuntao Li, Hanchu Zhang, Yutian Li, Sirui Wang, Wei Wu, Yan Zhang:
Pay More Attention to History: A Context Modeling Strategy for Conversational Text-to-SQL. 2718-2722 - Yuntao Li, Can Xu, Huang Hu, Lei Sha, Yan Zhang, Daxin Jiang:
Small Changes Make Big Differences: Improving Multi-turn Response Selection in Dialogue Systems via Fine-Grained Contrastive Learning. 2723-2727 - Marco Dinarelli, Marco Naguib, François Portet:
Toward Low-Cost End-to-End Spoken Language Understanding. 2728-2732 - Eleftherios Kapelonis, Efthymios Georgiou, Alexandros Potamianos:
A Multi-Task BERT Model for Schema-Guided Dialogue State Tracking. 2733-2737 - Heting Gao, Junrui Ni, Kaizhi Qian, Yang Zhang, Shiyu Chang, Mark Hasegawa-Johnson:
WavPrompt: Towards Few-Shot Spoken Language Understanding with Frozen Language Models. 2738-2742 - Asahi Ogushi, Toshiki Onishi, Yohei Tahara, Ryo Ishii, Atsushi Fukayama, Takao Nakamura, Akihiro Miyata:
Analysis of praising skills focusing on utterance contents. 2743-2747 - Pengwei Wang, Yinpei Su, Xiaohuan Zhou, Xin Ye, Liangchen Wei, Ming Liu, Yuan You, Feijun Jiang:
Speech2Slot: A Limited Generation Framework with Boundary Detection for Slot Filling from Speech. 2748-2752
Acoustic Signal Representation and Analysis I
- Khaled Koutini, Jan Schlüter, Hamid Eghbal-zadeh, Gerhard Widmer:
Efficient Training of Audio Transformers with Patchout. 2753-2757 - Mayank Sharma, Tarun Gupta, Kenny Qiu, Xiang Hao, Raffay Hamid:
CNN-based Audio Event Recognition for Automated Violence Classification and Rating for Prime Video Content. 2758-2762 - Hyeonuk Nam, Seong-Hu Kim, Byeong-Yun Ko, Yong-Hwa Park:
Frequency Dynamic Convolution: Frequency-Adaptive Pattern Recognition for Sound Event Detection. 2763-2767 - Zohreh Mostaani, Mathew Magimai-Doss:
On Breathing Pattern Information in Synthetic Speech. 2768-2772 - Chen Chen, Nana Hou, Yuchen Hu, Heqing Zou, Xiaofeng Qi, Eng Siong Chng:
Interactive Auido-text Representation for Automated Audio Captioning with Contrastive Learning. 2773-2777 - Yuya Yamamoto, Juhan Nam, Hiroko Terasawa:
Deformable CNN and Imbalance-Aware Feature Learning for Singing Technique Classification. 2778-2782
Privacy and Security in Speech Communication
- Nicolas M. Müller, Pavel Czempin, Franziska Dieckmann, Adam Froghyar, Konstantin Böttinger:
Does Audio Deepfake Detection Generalize? 2783-2787 - Nicolas M. Müller, Franziska Dieckmann, Jennifer Williams:
Attacker Attribution of Audio Deepfakes. 2788-2792 - Pierre Champion, Anthony Larcher, Denis Jouvet:
Are disentangled representations all you need to build speaker anonymization systems? 2793-2797 - Francisco Teixeira, Alberto Abad, Bhiksha Raj, Isabel Trancoso:
Towards End-to-End Private Automatic Speaker Recognition. 2798-2802 - Ehsan Amid, Om Dipakbhai Thakkar, Arun Narayanan, Rajiv Mathews, Françoise Beaufays:
Extracting Targeted Training Data from ASR Models, and How to Mitigate It. 2803-2807 - W. Ronny Huang, Steve Chien, Om Dipakbhai Thakkar, Rajiv Mathews:
Detecting Unintended Memorization in Language-Model-Fused ASR. 2808-2812
Multimodal Systems
- Shuta Taniguchi, Tsuneo Kato, Akihiro Tamura, Keiji Yasuda:
Transformer-Based Automatic Speech Recognition with Auxiliary Input of Source Language Text Toward Transcribing Simultaneous Interpretation. 2813-2817 - Valentin Gabeur, Paul Hongsuck Seo, Arsha Nagrani, Chen Sun, Karteek Alahari, Cordelia Schmid:
AVATAR: Unconstrained Audiovisual Speech Recognition. 2818-2822 - Puyuan Peng, David Harwath:
Word Discovery in Visually Grounded, Self-Supervised Speech Models. 2823-2827 - Richard Rose, Olivier Siohan:
End-to-End multi-talker audio-visual ASR using an active speaker attention module. 2828-2832 - Dmitriy Serdyuk, Otavio Braga, Olivier Siohan:
Transformer-Based Video Front-Ends for Audio-Visual Speech Recognition for Single and Muti-Person Video. 2833-2837 - Joanna Hong, Minsu Kim, Daehun Yoo, Yong Man Ro:
Visual Context-driven Audio Feature Enhancement for Robust End-to-End Audio-Visual Speech Recognition. 2838-2842
Atypical Speech Detection
- John B. Harvill, Mark Hasegawa-Johnson, Chang D. Yoo:
Frame-Level Stutter Detection. 2843-2847 - Darshana Priyasad, Andi Partovi, Sridha Sridharan, Maryam Kashefpoor, Tharindu Fernando, Simon Denman, Clinton Fookes, Jia Tang, David Kaye:
Detecting Heart Failure Through Voice Analysis using Self-Supervised Mode-Based Memory Fusion. 2848-2852 - Si Ioi Ng, Cymie Wing-Yee Ng, Jiarui Wang, Tan Lee:
Automatic Detection of Speech Sound Disorder in Child Speech Using Posterior-based Speaker Representations. 2853-2857 - Dominika Woszczyk, Anna Hlédiková, Alican Akman, Soteris Demetriou, Björn W. Schuller:
Data Augmentation for Dementia Detection in Spoken Language. 2858-2862 - Debottam Dutta, Debarpan Bhattacharya, Sriram Ganapathy, Amir Hossein Poorjam, Deepak Mittal, Maneesh Singh:
Acoustic Representation Learning on Breathing and Speech Signals for COVID-19 Detection. 2863-2867 - Sebastian Peter Bayerl, Dominik Wagner, Elmar Nöth, Korbinian Riedhammer:
Detecting Dysfluencies in Stuttering Therapy Using wav2vec 2.0. 2868-2872
Spoofing-Aware Automatic Speaker Verification (SASV) I
- Jeong-Hwan Choi, Joon-Young Yang, Ye-Rin Jeoung, Joon-Hyuk Chang:
HYU Submission for the SASV Challenge 2022: Reforming Speaker Embeddings with Spoofing-Aware Conditioning. 2873-2877 - Jungwoo Heo, Ju-Ho Kim, Hyun-seo Shin:
Two Methods for Spoofing-Aware Speaker Verification: Multi-Layer Perceptron Score Fusion Model and Integrated Embedding Projector. 2878-2882 - Chang Zeng, Lin Zhang, Meng Liu, Junichi Yamagishi:
Spoofing-Aware Attention based ASV Back-end with Multiple Enrollment Utterances and a Sampling Strategy for the SASV Challenge 2022. 2883-2887 - Alexander Alenin, Nikita Torgashov, Anton Okhotnikov, Rostislav Makarov, Ivan Yakovlev:
A Subnetwork Approach for Spoofing Aware Speaker Verification. 2888-2892 - Jee-weon Jung, Hemlata Tak, Hye-jin Shim, Hee-Soo Heo, Bong-Jin Lee, Soo-Whan Chung, Ha-Jin Yu, Nicholas W. D. Evans, Tomi Kinnunen:
SASV 2022: The First Spoofing-Aware Speaker Verification Challenge. 2893-2897 - Jin Woo Lee, Eungbeom Kim, Junghyun Koo, Kyogu Lee:
Representation Selective Self-distillation and wav2vec 2.0 Feature Exploration for Spoof-aware Speaker Verification. 2898-2902
Single-channel and multi-channel Speech Enhancement
- Nils L. Westhausen, Bernd T. Meyer:
tPLCnet: Real-time Deep Packet Loss Concealment in the Time Domain Using a Short Temporal Context. 2903-2907 - Kristina Tesch, Nils-Hendrik Mohrmann, Timo Gerkmann:
On the Role of Spatial, Spectral, and Temporal Processing for DNN-based Non-linear Multi-channel Speech Enhancement. 2908-2912 - Haoyu Li, Junichi Yamagishi:
DDS: A new device-degraded speech dataset for speech enhancement. 2913-2917 - Yicheng Du, Aditya Arie Nugraha, Kouhei Sekiguchi, Yoshiaki Bando, Mathieu Fontaine, Kazuyoshi Yoshii:
Direction-Aware Joint Adaptation of Neural Speech Enhancement and Recognition in Real Multiparty Conversational Environments. 2918-2922 - Julitta Bartolewska, Stanislaw Kacprzak, Konrad Kowalczyk:
Refining DNN-based Mask Estimation using CGMM-based EM Algorithm for Multi-channel Noise Reduction. 2923-2927 - Simon Welker, Julius Richter, Timo Gerkmann:
Speech Enhancement with Score-Based Generative Models in the Complex STFT Domain. 2928-2932 - Mohamed Nabih Ali, Alessio Brutti, Daniele Falavigna:
Enhancing Embeddings for Speech Classification in Noisy Conditions. 2933-2937 - Arnon Turetzky, Tzvi Michelson, Yossi Adi, Shmuel Peleg:
Deep Audio Waveform Prior. 2938-2942 - Mieszko Fras, Marcin Witkowski, Konrad Kowalczyk:
Convolutive Weighted Multichannel Wiener Filter Front-end for Distant Automatic Speech Recognition in Reverberant Multispeaker Scenarios. 2943-2947 - Danilo de Oliveira, Tal Peer, Timo Gerkmann:
Efficient Transformer-based Speech Enhancement Using Long Frames and STFT Magnitudes. 2948-2952 - Muqiao Yang, Joseph Konan, David Bick, Anurag Kumar, Shinji Watanabe, Bhiksha Raj:
Improving Speech Enhancement through Fine-Grained Speech Characteristics. 2953-2957
Voice Conversion and Adaptation II
- Piotr Bilinski, Thomas Merritt, Abdelhamid Ezzerg, Kamil Pokora, Sebastian Cygert, Kayoko Yanagisawa, Roberto Barra-Chicote, Daniel Korzekwa:
Creating New Voices using Normalizing Flows. 2958-2962 - Ariadna Sánchez, Alessio Falai, Ziyao Zhang, Orazio Angelini, Kayoko Yanagisawa:
Unify and Conquer: How Phonetic Feature Representation Affects Polyglot Text-To-Speech (TTS). 2963-2967 - Kenta Udagawa, Yuki Saito, Hiroshi Saruwatari:
Human-in-the-loop Speaker Adaptation for DNN-based Multi-speaker TTS. 2968-2972 - Magdalena Proszewska, Grzegorz Beringer, Daniel Sáez-Trigueros, Thomas Merritt, Abdelhamid Ezzerg, Roberto Barra-Chicote:
GlowVC: Mel-spectrogram space disentangling model for language-independent text-free voice conversion. 2973-2977 - Jaeuk Lee, Joon-Hyuk Chang:
One-Shot Speaker Adaptation Based on Initialization by Generative Adversarial Networks for TTS. 2978-2982 - Alon Levkovitch, Eliya Nachmani, Lior Wolf:
Zero-Shot Voice Conditioning for Denoising Diffusion TTS Models. 2983-2987 - Jaeuk Lee, Joon-Hyuk Chang:
Advanced Speaker Embedding with Predictive Variance of Gaussian Distribution for Speaker Adaptation in TTS. 2988-2992 - Panagiotis Kakoulidis, Nikolaos Ellinas, Georgios Vamvoukakis, Konstantinos Markopoulos, June Sig Sung, Gunu Jho, Pirros Tsiakoulis, Aimilios Chalamandaris:
Karaoker: Alignment-free singing voice synthesis with speech training data. 2993-2997 - Ji Sub Um, Yeunju Choi, Hoi Rin Kim:
ACNN-VC: Utilizing Adaptive Convolution Neural Network for One-Shot Voice Conversion. 2998-3002 - Tasnima Sadekova, Vladimir Gogoryan, Ivan Vovk, Vadim Popov, Mikhail A. Kudinov, Jiansheng Wei:
A Unified System for Voice Cloning and Voice Conversion through Diffusion Probabilistic Modeling. 3003-3007 - Tae-Woo Kim, Min-Su Kang, Gyeong-Hoon Lee:
Adversarial Multi-Task Learning for Disentangling Timbre and Pitch in Singing Voice Synthesis. 3008-3012 - Shrutina Agarwal, Naoya Takahashi, Sriram Ganapathy:
Leveraging Symmetrical Convolutional Transformer Networks for Speech to Singing Voice Style Transfer. 3013-3017