17,853 research outputs found
Summarization of Spanish Talk Shows with Siamese Hierarchical Attention Networks
[EN] In this paper, we present an approach to Spanish talk shows summarization. Our approach is based on the use of Siamese Neural Networks on the transcription of the show audios. Specifically, we propose to use Hierarchical Attention Networks to select the most relevant sentences for each speaker about a given topic in the show, in order to summarize his opinion about the topic. We train these networks in a siamese way to determine whether a summary is appropriate or not. Previous evaluation of this approach on summarization task of English newspapers achieved performances similar to other state-of-the-art systems. In the absence of enough transcribed or recognized speech data to train our system for talk show summarization in Spanish, we acquire a large corpus of document-summary pairs from Spanish newspapers and we use it to train our system. We choose this newspapers domain due to its high similarity with the topics addressed in talk shows. A preliminary evaluation of our summarization system on Spanish TV programs shows the adequacy of the proposal.This work has been partially supported by the Spanish MINECO and FEDER founds under project AMIC (TIN2017-85854-C4-2-R). Work of Jose-Angel Gonzalez is financed by Universitat Politecnica de Valencia under grant PAID-01-17.González-Barba, JÁ.; Hurtado Oliver, LF.; Segarra Soriano, E.; García-Granada, F.; Sanchís Arnal, E. (2019). Summarization of Spanish Talk Shows with Siamese Hierarchical Attention Networks. Applied Sciences. 9(18):1-13. https://doi.org/10.3390/app9183836S113918Carbonell, J., & Goldstein, J. (1998). The use of MMR, diversity-based reranking for reordering documents and producing summaries. Proceedings of the 21st annual international ACM SIGIR conference on Research and development in information retrieval - SIGIR ’98. doi:10.1145/290941.291025Erkan, G., & Radev, D. R. (2004). LexRank: Graph-based Lexical Centrality as Salience in Text Summarization. Journal of Artificial Intelligence Research, 22, 457-479. doi:10.1613/jair.1523Lloret, E., & Palomar, M. (2011). Text summarisation in progress: a literature review. Artificial Intelligence Review, 37(1), 1-41. doi:10.1007/s10462-011-9216-zSee, A., Liu, P. J., & Manning, C. D. (2017). Get To The Point: Summarization with Pointer-Generator Networks. Proceedings of the 55th Annual Meeting of the Association for
Computational Linguistics (Volume 1: Long Papers). doi:10.18653/v1/p17-1099Narayan, S., Cohen, S. B., & Lapata, M. (2018). Ranking Sentences for Extractive Summarization with Reinforcement Learning. Proceedings of the 2018 Conference of the North American Chapter of
the Association for Computational Linguistics: Human Language
Technologies, Volume 1 (Long Papers). doi:10.18653/v1/n18-1158González, J.-Á., Segarra, E., García-Granada, F., Sanchis, E., & Hurtado, L.-F. (2019). Siamese hierarchical attention networks for extractive summarization. Journal of Intelligent & Fuzzy Systems, 36(5), 4599-4607. doi:10.3233/jifs-179011Furui, S., Kikuchi, T., Shinnaka, Y., & Hori, C. (2004). Speech-to-Text and Speech-to-Speech Summarization of Spontaneous Speech. IEEE Transactions on Speech and Audio Processing, 12(4), 401-408. doi:10.1109/tsa.2004.828699Shih-Hung Liu, Kuan-Yu Chen, Chen, B., Hsin-Min Wang, Hsu-Chun Yen, & Wen-Lian Hsu. (2015). Combining Relevance Language Modeling and Clarity Measure for Extractive Speech Summarization. IEEE/ACM Transactions on Audio, Speech, and Language Processing, 23(6), 957-969. doi:10.1109/taslp.2015.2414820Yang, Z., Yang, D., Dyer, C., He, X., Smola, A., & Hovy, E. (2016). Hierarchical Attention Networks for Document Classification. Proceedings of the 2016 Conference of the North American Chapter of the Association for Computational Linguistics: Human Language Technologies. doi:10.18653/v1/n16-1174Conneau, A., Kiela, D., Schwenk, H., Barrault, L., & Bordes, A. (2017). Supervised Learning of Universal Sentence Representations from
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TasNet: time-domain audio separation network for real-time, single-channel speech separation
Robust speech processing in multi-talker environments requires effective
speech separation. Recent deep learning systems have made significant progress
toward solving this problem, yet it remains challenging particularly in
real-time, short latency applications. Most methods attempt to construct a mask
for each source in time-frequency representation of the mixture signal which is
not necessarily an optimal representation for speech separation. In addition,
time-frequency decomposition results in inherent problems such as
phase/magnitude decoupling and long time window which is required to achieve
sufficient frequency resolution. We propose Time-domain Audio Separation
Network (TasNet) to overcome these limitations. We directly model the signal in
the time-domain using an encoder-decoder framework and perform the source
separation on nonnegative encoder outputs. This method removes the frequency
decomposition step and reduces the separation problem to estimation of source
masks on encoder outputs which is then synthesized by the decoder. Our system
outperforms the current state-of-the-art causal and noncausal speech separation
algorithms, reduces the computational cost of speech separation, and
significantly reduces the minimum required latency of the output. This makes
TasNet suitable for applications where low-power, real-time implementation is
desirable such as in hearable and telecommunication devices.Comment: Camera ready version for ICASSP 2018, Calgary, Canad
Direction of Arrival with One Microphone, a few LEGOs, and Non-Negative Matrix Factorization
Conventional approaches to sound source localization require at least two
microphones. It is known, however, that people with unilateral hearing loss can
also localize sounds. Monaural localization is possible thanks to the
scattering by the head, though it hinges on learning the spectra of the various
sources. We take inspiration from this human ability to propose algorithms for
accurate sound source localization using a single microphone embedded in an
arbitrary scattering structure. The structure modifies the frequency response
of the microphone in a direction-dependent way giving each direction a
signature. While knowing those signatures is sufficient to localize sources of
white noise, localizing speech is much more challenging: it is an ill-posed
inverse problem which we regularize by prior knowledge in the form of learned
non-negative dictionaries. We demonstrate a monaural speech localization
algorithm based on non-negative matrix factorization that does not depend on
sophisticated, designed scatterers. In fact, we show experimental results with
ad hoc scatterers made of LEGO bricks. Even with these rudimentary structures
we can accurately localize arbitrary speakers; that is, we do not need to learn
the dictionary for the particular speaker to be localized. Finally, we discuss
multi-source localization and the related limitations of our approach.Comment: This article has been accepted for publication in IEEE/ACM
Transactions on Audio, Speech, and Language processing (TASLP
Efficient Synthesis of Room Acoustics via Scattering Delay Networks
An acoustic reverberator consisting of a network of delay lines connected via
scattering junctions is proposed. All parameters of the reverberator are
derived from physical properties of the enclosure it simulates. It allows for
simulation of unequal and frequency-dependent wall absorption, as well as
directional sources and microphones. The reverberator renders the first-order
reflections exactly, while making progressively coarser approximations of
higher-order reflections. The rate of energy decay is close to that obtained
with the image method (IM) and consistent with the predictions of Sabine and
Eyring equations. The time evolution of the normalized echo density, which was
previously shown to be correlated with the perceived texture of reverberation,
is also close to that of IM. However, its computational complexity is one to
two orders of magnitude lower, comparable to the computational complexity of a
feedback delay network (FDN), and its memory requirements are negligible
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