Google
Google Scholar
Semantic Scholar
IA Scholar
CiteSeerX
ORCIDIEEE/ACM Transactions on Audio, Speech and Language Processing, Volume 24
Volume 24, Number 1, January 2016
- Sandrine Brognaux, Thomas Drugman:
HMM-Based Speech Segmentation: Improvements of Fully Automatic Approaches. 5-15
- Marie Tahon
, Laurence Devillers:
Towards a Small Set of Robust Acoustic Features for Emotion Recognition: Challenges. 16-28 - Hamid Behravan, Ville Hautamäki, Sabato Marco Siniscalchi, Tomi Kinnunen, Chin-Hui Lee:
i-Vector Modeling of Speech Attributes for Automatic Foreign Accent Recognition. 29-41 - Rahim Saeidi, Paavo Alku, Tom Bäckström:
Feature Extraction Using Power-Law Adjusted Linear Prediction With Application to Speaker Recognition Under Severe Vocal Effort Mismatch. 42-53 - Iman Tabatabaei Ardekani, Jari P. Kaipio, Alireza Nasiri, Hamid R. Sharifzadeh, Waleed H. Abdulla:
A Statistical Inverse Problem Approach to Online Secondary Path Modeling in Active Noise Control. 54-64 - Themos Stafylakis, Patrick Kenny, Md. Jahangir Alam, Marcel Kockmann:
Speaker and Channel Factors in Text-Dependent Speaker Recognition. 65-78 - Yanzhang He, Peter Baumann, Hao Fang, Brian Hutchinson, Aaron Jaech, Mari Ostendorf, Eric Fosler-Lussier, Janet B. Pierrehumbert:
Using Pronunciation-Based Morphological Subword Units to Improve OOV Handling in Keyword Search. 79-92 - Meng Sun, Xiongwei Zhang, Hugo Van hamme, Thomas Fang Zheng:
Unseen Noise Estimation Using Separable Deep Auto Encoder for Speech Enhancement. 93-104 - Luciana Ferrer, Yun Lei, Mitchell McLaren, Nicolas Scheffer:
Study of Senone-Based Deep Neural Network Approaches for Spoken Language Recognition. 105-116 - Stefan Ingi Adalbjornsson, Ted Kronvall, Simon Burgess, Kalle Åström, Andreas Jakobsson:
Sparse Localization of Harmonic Audio Sources. 117-129 - Man-Wai Mak, Xiaomin Pang, Jen-Tzung Chien:
Mixture of PLDA for Noise Robust I-Vector Speaker Verification. 130-142 - Craig A. Anderson, Paul D. Teal, Mark A. Poletti:
Spatial Correlation of Radial Gaussian and Uniform Spherical Volume Near-Field Source Distributions. 143-150 - Humberto M. Torres, Jorge A. Gurlekian:
Novel Estimation Method for the Superpositional Intonation Model. 151-160 - Stefan Bilbao, Brian Hamilton, Jonathan Botts, Lauri Savioja:
Finite Volume Time Domain Room Acoustics Simulation under General Impedance Boundary Conditions. 161-173 - Amir Hossein Harati Nejad Torbati, Joseph Picone:
A Doubly Hierarchical Dirichlet Process Hidden Markov Model with a Non-Ergodic Structure. 174-184 - Jen-Tzung Chien, Po-Kai Yang:
Bayesian Factorization and Learning for Monaural Source Separation. 185-195 - David Lou Alon, Boaz Rafaely:
Beamforming with Optimal Aliasing Cancellation in Spherical Microphone Arrays. 196-210
Volume 24, Number 2, February 2016
- Eugen Rasumow, Martin Hansen, Steven van de Par, Dirk Puschel, Volker Mellert, Simon Doclo, Matthias Blau:
Regularization Approaches for Synthesizing HRTF Directivity Patterns. 215-225 - Chao Pan, Jacob Benesty, Jingdong Chen:
Design of Directivity Patterns with a Unique Null of Maximum Multiplicity. 226-235 - Jeih-Weih Hung, Hsin-Ju Hsieh, Berlin Chen:
Robust Speech Recognition via Enhancing the Complex-Valued Acoustic Spectrum in Modulation Domain. 236-251 - Xiao-Lei Zhang, DeLiang Wang:
Boosting Contextual Information for Deep Neural Network Based Voice Activity Detection. 252-264 - M. A. Tugtekin Turan, Engin Erzin:
Source and Filter Estimation for Throat-Microphone Speech Enhancement. 265-275 - Nasser Mohammadiha, Simon Doclo:
Speech Dereverberation Using Non-Negative Convolutive Transfer Function and Spectro-Temporal Modeling. 276-289 - Anil Sharma, Sanjit K. Kaul:
Two-Stage Supervised Learning-Based Method to Detect Screams and Cries in Urban Environments. 290-299 - Xiaoguang Wu, Huawei Chen:
Directivity Factors of the First-Order Steerable Differential Array With Microphone Mismatches: Deterministic and Worst-Case Analysis. 300-315 - Andreas I. Koutrouvelis, George P. Kafentzis, Nikolay D. Gaubitch, Richard Heusdens:
A Fast Method for High-Resolution Voiced/Unvoiced Detection and Glottal Closure/Opening Instant Estimation of Speech. 316-328 - Tomohiko Nakamura, Eita Nakamura, Shigeki Sagayama:
Real-Time Audio-to-Score Alignment of Music Performances Containing Errors and Arbitrary Repeats and Skips. 329-339 - Adrian Bahne, Anders Ahlén:
Optimizing the Similarity of Loudspeaker-Room Responses in Multiple Listening Positions. 340-353 - James M. Kates, Kathryn Hoberg Arehart:
The Hearing-Aid Audio Quality Index (HAAQI). 354-365 - Henning F. Schepker, Simon Doclo:
A Semidefinite Programming Approach to Min-max Estimation of the Common Part of Acoustic Feedback Paths in Hearing Aids. 366-377 - Bong-Ki Lee, Joon-Hyuk Chang:
Packet Loss Concealment Based on Deep Neural Networks for Digital Speech Transmission. 378-387 - Luisa Bentivogli, Nicola Bertoldi, Mauro Cettolo, Marcello Federico, Matteo Negri, Marco Turchi:
On the Evaluation of Adaptive Machine Translation for Human Post-Editing. 388-399
Volume 24, Number 3, March 2016
- Reinhard Sonnleitner, Gerhard Widmer:
Robust Quad-Based Audio Fingerprinting. 409-421 - Li Dong, Furu Wei, Ke Xu, Shixia Liu, Ming Zhou:
Adaptive Multi-Compositionality for Recursive Neural Network Models. 422-431 - Zheng Lin, Xiaolong Jin, Xueke Xu, Yuanzhuo Wang, Xueqi Cheng, Weiping Wang, Dan Meng:
An Unsupervised Cross-Lingual Topic Model Framework for Sentiment Classification. 432-444 - Anil M. Nagathil, Claus Weihs, Rainer Martin:
Spectral Complexity Reduction of Music Signals for Mitigating Effects of Cochlear Hearing Loss. 445-458 - Tian Tan, Yanmin Qian, Kai Yu:
Cluster Adaptive Training for Deep Neural Network Based Acoustic Model. 459-468 - Leijon Leijon, Gustav Eje Henter, Martin Dahlquist:
Bayesian Analysis of Phoneme Confusion Matrices. 469-482 - Donald S. Williamson, Yuxuan Wang, DeLiang Wang:
Complex Ratio Masking for Monaural Speech Separation. 483-492 - Johannes Traa, David Wingate, Noah D. Stein, Paris Smaragdis:
Robust Source Localization and Enhancement With a Probabilistic Steered Response Power Model. 493-503 - Sven Ewan Shepstone, Kong-Aik Lee, Haizhou Li, Zheng-Hua Tan, Søren Holdt Jensen:
Total Variability Modeling Using Source-Specific Priors. 504-517 - Martin Schneider, Walter Kellermann:
Multichannel Acoustic Echo Cancellation in the Wave Domain With Increased Robustness to Nonuniqueness. 518-529 - Ken O'Hanlon, Hidehisa Nagano, Nicolas Keriven, Mark D. Plumbley:
Non-Negative Group Sparsity with Subspace Note Modelling for Polyphonic Transcription. 530-542 - Elior Hadad, Simon Doclo, Sharon Gannot:
The Binaural LCMV Beamformer and its Performance Analysis. 543-558 - Felipe Grijalva, Luiz Martini, Dinei Florêncio, Siome Goldenstein:
A Manifold Learning Approach for Personalizing HRTFs from Anthropometric Features. 559-570 - Lin Wang, Simon Doclo:
Correlation Maximization-Based Sampling Rate Offset Estimation for Distributed Microphone Arrays. 571-582 - Nasim Radmanesh, Ian S. Burnett, Bhaskar D. Rao:
A Lasso-LS Optimization with a Frequency Variable Dictionary in a Multizone Sound System. 583-593 - Xin Liu, Changchun Bao:
Audio Bandwidth Extension Based on Ensemble Echo State Networks with Temporal Evolution. 594-607
Volume 24, Number 4, April 2016
- Peifeng Li, Guodong Zhou:
Joint Argument Inference in Chinese Event Extraction with Argument Consistency and Event Relevance. 612-622 - Jianming Liu, Steven L. Grant:
Proportionate Adaptive Filtering for Block-Sparse System Identification. 623-630 - Jesper Rindom Jensen, Jacob Benesty, Mads Græsbøll Christensen:
Noise Reduction with Optimal Variable Span Linear Filters. 631-644 - Sidsel Marie Nørholm, Jesper Rindom Jensen, Mads Græsbøll Christensen:
Enhancement and Noise Statistics Estimation for Non-Stationary Voiced Speech. 645-658 - Daryush D. Mehta, Jarrad H. Van Stan, Robert E. Hillman:
Relationships Between Vocal Function Measures Derived from an Acoustic Microphone and a Subglottal Neck-Surface Accelerometer. 659-668 - Herman Kamper, Aren Jansen, Sharon Goldwater:
Unsupervised Word Segmentation and Lexicon Discovery Using Acoustic Word Embeddings. 669-679 - Ina Kodrasi, Simon Doclo:
Joint Dereverberation and Noise Reduction Based on Acoustic Multi-Channel Equalization. 680-693 - Hamid Palangi, Li Deng, Yelong Shen, Jianfeng Gao, Xiaodong He, Jianshu Chen, Xinying Song, Rabab K. Ward:
Deep Sentence Embedding Using Long Short-Term Memory Networks: Analysis and Application to Information Retrieval. 694-707 - Michael Jeffet, Noam R. Shabtai, Boaz Rafaely:
Theory and Perceptual Evaluation of the Binaural Reproduction and Beamforming Tradeoff in the Generalized Spherical Array Beamformer. 708-718 - Pablo Peso Parada, Dushyant Sharma, Jose Lainez, Daniel Barreda, Toon van Waterschoot, Patrick A. Naylor:
A Single-Channel Non-Intrusive C50 Estimator Correlated With Speech Recognition Performance. 719-732 - Ming-Hsiang Su, Chung-Hsien Wu, Yu-Ting Zheng:
Exploiting Turn-Taking Temporal Evolution for Personality Trait Perception in Dyadic Conversations. 733-744 - Sadaf Abdul-Rauf, Holger Schwenk, Patrik Lambert, Mohammad Nawaz:
Empirical Use of Information Retrieval to Build Synthetic Data for SMT Domain Adaptation. 745-754 - Shinnosuke Takamichi, Tomoki Toda, Alan W. Black, Graham Neubig, Sakriani Sakti, Satoshi Nakamura:
Postfilters to Modify the Modulation Spectrum for Statistical Parametric Speech Synthesis. 755-767 - Zhizheng Wu, Phillip L. De Leon, Cenk Demiroglu, Ali Khodabakhsh, Simon King, Zhen-Hua Ling, Daisuke Saito, Bryan Stewart, Tomoki Toda, Mirjam Wester, Junichi Yamagishi:
Anti-Spoofing for Text-Independent Speaker Verification: An Initial Database, Comparison of Countermeasures, and Human Performance. 768-783 - Kristian Timm Andersen, Marc Moonen:
Adaptive Time-Frequency Analysis for Noise Reduction in an Audio Filter Bank With Low Delay. 784-795 - Zhong-Qiu Wang, DeLiang Wang:
A Joint Training Framework for Robust Automatic Speech Recognition. 796-806 - Huy Phan, Lars Hertel, Marco Maaß, Radoslaw Mazur, Alfred Mertins:
Learning Representations for Nonspeech Audio Events Through Their Similarities to Speech Patterns. 807-822
Volume 24, Number 5, May 2016
- T. J. Tsai, Andreas Stolcke:
Robust and Efficient Multiple Alignment of Unsynchronized Meeting Recordings. 833-845 - Simon Receveur, Robin Weib, Tim Fingscheidt:
Turbo Automatic Speech Recognition. 846-862 - Ricard Marxer, Hendrik Purwins:
Unsupervised Incremental Online Learning and Prediction of Musical Audio Signals. 863-874 - Mohammad Adeli, Jean Rouat, Sean U. N. Wood, Stephane Molotchnikoff, Eric Plourde:
A Flexible Bio-Inspired Hierarchical Model for Analyzing Musical Timbre. 875-889 - Geliang Zhang, Simon J. Godsill:
Fundamental Frequency Estimation in Speech Signals With Variable Rate Particle Filters. 890-900 - Nadine Kroher, Emilia Gómez:
Automatic Transcription of Flamenco Singing From Polyphonic Music Recordings. 901-913 - Fiete Winter, Jens Ahrens, Sascha Spors:
On Analytic Methods for 2.5-D Local Sound Field Synthesis Using Circular Distributions of Secondary Sources. 914-926 - Siddharth Sigtia, Emmanouil Benetos, Simon Dixon:
An End-to-End Neural Network for Polyphonic Piano Music Transcription. 927-939 - Martin Krawczyk-Becker, Timo Gerkmann:
Fundamental Frequency Informed Speech Enhancement in a Flexible Statistical Framework. 940-951 - Joseph Szurley, Alexander Bertrand, Bas van Dijk, Marc Moonen:
Binaural Noise Cue Preservation in a Binaural Noise Reduction System With a Remote Microphone Signal. 952-966 - Xiao-Lei Zhang, DeLiang Wang:
A Deep Ensemble Learning Method for Monaural Speech Separation. 967-977 - Haotian Xu, Zhijian Ou:
Scalable Discovery of Audio Fingerprint Motifs in Broadcast Streams With Determinantal Point Process Based Motif Clustering. 978-989
Volume 24, Number 6, June 2016
- Asli Celikyilmaz, Ruhi Sarikaya, Minwoo Jeong, Anoop Deoras:
An Empirical Investigation of Word Class-Based Features for Natural Language Understanding. 994-1005 - Duc Hoang Ha Nguyen, Xiong Xiao, Eng Siong Chng, Haizhou Li:
Feature Adaptation Using Linear Spectro-Temporal Transform for Robust Speech Recognition. 1006-1019 - Xiaojun Qian, Helen M. Meng, Frank K. Soong:
A Two-Pass Framework of Mispronunciation Detection and Diagnosis for Computer-Aided Pronunciation Training. 1020-1028 - Lijiang Chen, Xia Mao, Hong Yan:
Text-Independent Phoneme Segmentation Combining EGG and Speech Data. 1029-1037 - Vincent Mohammad Tavakoli, Jesper Rindom Jensen, Mads Græsbøll Christensen, Jacob Benesty:
A Framework for Speech Enhancement With Ad Hoc Microphone Arrays. 1038-1051 - Yan-You Chen, Chung-Hsien Wu, Yi-Chin Huang, Shih-Lun Lin, Jhing-Fa Wang:
Candidate Expansion and Prosody Adjustment for Natural Speech Synthesis Using a Small Corpus. 1052-1065 - Xueliang Zhang, Hui Zhang, Shuai Nie, Guanglai Gao, Wenju Liu:
A Pairwise Algorithm Using the Deep Stacking Network for Speech Separation and Pitch Estimation. 1066-1078 - Lin Wang, Tsz-Kin Hon, Joshua D. Reiss, Andrea Cavallaro:
An Iterative Approach to Source Counting and Localization Using Two Distant Microphones. 1079-1093 - Seán O'Leary, Axel Röbel:
A Montage Approach to Sound Texture Synthesis. 1094-1105 - Chahid Ouali, Pierre Dumouchel, Vishwa Gupta:
Fast Audio Fingerprinting System Using GPU and a Clustering-Based Technique. 1106-1118 - Francisco Raposo, Ricardo Ribeiro, David Martins de Matos:
Using Generic Summarization to Improve Music Information Retrieval Tasks. 1119-1128 - Lantian Li, Dong Wang, Chenhao Zhang, Thomas Fang Zheng:
Improving Short Utterance Speaker Recognition by Modeling Speech Unit Classes. 1129-1139 - Jalal Taghia, Rainer Martin:
A Frequency-Domain Adaptive Line Enhancer With Step-Size Control Based on Mutual Information for Harmonic Noise Reduction. 1140-1154
Volume 24, Number 7, July 2016
- Min Gao, Jing Lu, Xiaojun Qiu:
A Simplified Subband ANC Algorithm Without Secondary Path Modeling. 1164-1174 - Ryo Aihara, Tetsuya Takiguchi, Yasuo Ariki:
Multiple Non-Negative Matrix Factorization for Many-to-Many Voice Conversion. 1175-1184 - Kai Chen, Qiang Huo:
Training Deep Bidirectional LSTM Acoustic Model for LVCSR by a Context-Sensitive-Chunk BPTT Approach. 1185-1193 - Themos Stafylakis, Md. Jahangir Alam, Patrick Kenny:
Text-Dependent Speaker Recognition With Random Digit Strings. 1194-1203 - K. T. Deepak, S. R. Mahadeva Prasanna:
Foreground Speech Segmentation and Enhancement Using Glottal Closure Instants and Mel Cepstral Coefficients. 1204-1218 - Habib Hajimolahoseini, Rassoul Amirfattahi, Saeed Gazor, Hamid Soltanian-Zadeh:
Robust Estimation and Tracking of Pitch Period Using an Efficient Bayesian Filter. 1219-1229 - Subhasmita Sahoo, Aurobinda Routray:
A Novel Method of Glottal Inverse Filtering. 1230-1241 - Gilles Degottex, Luc Ardaillon, Axel Roebel:
Multi-Frame Amplitude Envelope Estimation for Modification of Singing Voice. 1242-1254 - Zhizheng Wu, Simon King:
Improving Trajectory Modelling for DNN-Based Speech Synthesis by Using Stacked Bottleneck Features and Minimum Generation Error Training. 1255-1265 - Xabier Jaureguiberry, Emmanuel Vincent, Gaël Richard:
Fusion Methods for Speech Enhancement and Audio Source Separation. 1266-1279 - Rajib Lochan Das, Mrityunjoy Chakraborty:
Improving the Performance of the PNLMS Algorithm Using l1 Norm Regularization. 1280-1290 - Maja Taseska, Emanuël A. P. Habets:
Spotforming: Spatial Filtering With Distributed Arrays for Position-Selective Sound Acquisition. 1291-1304 - Guangyou Zhou, Zhiwen Xie, Tingting He, Jun Zhao, Xiaohua Tony Hu:
Learning the Multilingual Translation Representations for Question Retrieval in Community Question Answering via Non-Negative Matrix Factorization. 1305-1314 - Chanwoo Kim, Richard M. Stern:
Power-Normalized Cepstral Coefficients (PNCC) for Robust Speech Recognition. 1315-1329
Volume 24, Number 8, August 2016
- Henning F. Schepker, Simon Doclo:
Least-Squares Estimation of the Common Pole-Zero Filter of Acoustic Feedback Paths in Hearing Aids. 1334-1347 - Hannes Pessentheiner, Martin Hagmüller, Gernot Kubin:
Localization and Characterization of Multiple Harmonic Sources. 1348-1363 - Hanieh Khalilian, Ivan V. Bajic, Rodney G. Vaughan:
Comparison of Loudspeaker Placement Methods for Sound Field Reproduction. 1364-1379 - Cheng-Yen Yang, Chih-Wei Liu, Shyh-Jye Jou:
A Systematic ANSI S1.11 Filter Bank Specification Relaxation and Its Efficient Multirate Architecture for Hearing-Aid Systems. 1380-1392 - Bracha Laufer-Goldshtein, Ronen Talmon, Sharon Gannot:
Semi-Supervised Sound Source Localization Based on Manifold Regularization. 1393-1407 - Dionyssos Kounades-Bastian, Laurent Girin, Xavier Alameda-Pineda, Sharon Gannot, Radu Horaud:
A Variational EM Algorithm for the Separation of Time-Varying Convolutive Audio Mixtures. 1408-1423 - Jun Du, Yanhui Tu, Li-Rong Dai, Chin-Hui Lee:
A Regression Approach to Single-Channel Speech Separation Via High-Resolution Deep Neural Networks. 1424-1437 - Xunying Liu, Xie Chen, Yongqiang Wang, Mark J. F. Gales, Philip C. Woodland:
Two Efficient Lattice Rescoring Methods Using Recurrent Neural Network Language Models. 1438-1449 - Pawel Swietojanski, Jinyu Li, Steve Renals:
Learning Hidden Unit Contributions for Unsupervised Acoustic Model Adaptation. 1450-1463 - Meng Zhang, Yang Liu, Huanbo Luan, Maosong Sun:
Listwise Ranking Functions for Statistical Machine Translation. 1464-1472
Volume 24, Number 9, September 2016
- Daniel Cruz Cavalieri, Sira E. Palazuelos-Cagigas, Teodiano Freire Bastos Filho, Mário Sarcinelli Filho:
Combination of Language Models for Word Prediction: An Exponential Approach. 1481-1494 - Ofer Schwartz, Sharon Gannot, Emanuël A. P. Habets:
An Expectation-Maximization Algorithm for Multimicrophone Speech Dereverberation and Noise Reduction With Coherence Matrix Estimation. 1495-1510 - Symeon Delikaris-Manias, Juha Vilkamo, Ville Pulkki:
Signal-Dependent Spatial Filtering Based on Weighted-Orthogonal Beamformers in the Spherical Harmonic Domain. 1511-1523 - Sheng Li, Yuya Akita, Tatsuya Kawahara:
Semi-Supervised Acoustic Model Training by Discriminative Data Selection From Multiple ASR Systems' Hypotheses. 1524-1534 - Christian Dittmar, Meinard Müller:
Reverse Engineering the Amen Break - Score-Informed Separation and Restoration Applied to Drum Recordings. 1535-1547 - Chao Pan, Jingdong Chen, Jacob Benesty:
Reduced-Order Robust Superdirective Beamforming With Uniform Linear Microphone Arrays. 1548-1559 - Derry Fitzgerald, Antoine Liutkus, Roland Badeau:
Projection-Based Demixing of Spatial Audio. 1560-1572 - Lin Wang, Joshua D. Reiss, Andrea Cavallaro:
Over-Determined Source Separation and Localization Using Distributed Microphones. 1573-1588 - Yang Liu, Sujian Li, Furu Wei, Heng Ji:
Relation Classification Via Modeling Augmented Dependency Paths. 1589-1598 - Adam Kuklasinski, Simon Doclo, Søren Holdt Jensen, Jesper Jensen:
Maximum Likelihood PSD Estimation for Speech Enhancement in Reverberation and Noise. 1599-1612 - Sam Karimian-Azari, Jesper Rindom Jensen, Mads Græsbøll Christensen:
Computationally Efficient and Noise Robust DOA and Pitch Estimation. 1613-1625 - Daichi Kitamura, Nobutaka Ono, Hiroshi Sawada, Hirokazu Kameoka, Hiroshi Saruwatari:
Determined Blind Source Separation Unifying Independent Vector Analysis and Nonnegative Matrix Factorization. 1626-1641 - Nicolas Obin, Axel Roebel:
Similarity Search of Acted Voices for Automatic Voice Casting. 1642-1651 - Aditya Arie Nugraha, Antoine Liutkus, Emmanuel Vincent:
Multichannel Audio Source Separation With Deep Neural Networks. 1652-1664 - Stephen H. Shum, David F. Harwath, Najim Dehak, James R. Glass:
On the Use of Acoustic Unit Discovery for Language Recognition. 1665-1676
Volume 24, Number 10, October 2016
- James Eaton, Nikolay D. Gaubitch, Alastair H. Moore, Patrick A. Naylor:
Estimation of Room Acoustic Parameters: The ACE Challenge. 1681-1693 - Takashi Nose:
Efficient Implementation of Global Variance Compensation for Parametric Speech Synthesis. 1694-1704 - Shabnam Ghaffarzadegan, Hynek Boril, John H. L. Hansen:
Generative Modeling of Pseudo-Whisper for Robust Whispered Speech Recognition. 1705-1720 - Seyedmahdad Mirsamadi, John H. L. Hansen:
A Generalized Nonnegative Tensor Factorization Approach for Distant Speech Recognition With Distributed Microphones. 1721-1731 - Laura Fuster, Maria de Diego, Luis Antonio Azpicueta-Ruiz