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Block-online Multi-channel Speech Enhancement Using Deep Neural Network-supported Relative Transfer Function Estimates

The result's identifiers

  • Result code in IS VaVaI

    <a href="https://www.isvavai.cz/riv?ss=detail&h=RIV%2F46747885%3A24220%2F20%3A00006552" target="_blank" >RIV/46747885:24220/20:00006552 - isvavai.cz</a>

  • Result on the web

    <a href="https://digital-library.theiet.org/content/journals/10.1049/iet-spr.2019.0304" target="_blank" >https://digital-library.theiet.org/content/journals/10.1049/iet-spr.2019.0304</a>

  • DOI - Digital Object Identifier

    <a href="http://dx.doi.org/10.1049/iet-spr.2019.0304" target="_blank" >10.1049/iet-spr.2019.0304</a>

Alternative languages

  • Result language

    angličtina

  • Original language name

    Block-online Multi-channel Speech Enhancement Using Deep Neural Network-supported Relative Transfer Function Estimates

  • Original language description

    This work addresses the problem of block-online processing for multi-channel speech enhancement. Such processing is vital in scenarios with moving speakers and/or when very short utterances are processed, e.g., in voice assistant scenarios. We consider several variants of a system that performs beamforming supported by DNN-based voice activity detection (VAD) followed by post-filtering. The speaker is targeted through estimating relative transfer functions between microphones. Each block of the input signals is processed independently in order to make the method applicable in highly dynamic environments. Owing to the short length of the processed block, the statistics required by the beamformer are estimated less precisely. The influence of this inaccuracy is studied and compared to the processing regime when recordings are treated as one block (batch processing). The experimental evaluation of the proposed method is performed on large datasets of CHiME-4 and on another dataset featuring moving target speaker. The experiments are evaluated in terms of objective and perceptual criteria (such as signal-to-interference ratio (SIR) or perceptual evaluation of speech quality (PESQ), respectively). Moreover, word error rate (WER) achieved by a baseline automatic speech recognition system is evaluated, for which the enhancement method serves as a front-end solution. The results indicate that the proposed method is robust with respect to short length of the processed block. Significant improvements in terms of the criteria and WER are observed even for the block length of 250 ms.

  • Czech name

  • Czech description

Classification

  • Type

    J<sub>imp</sub> - Article in a specialist periodical, which is included in the Web of Science database

  • CEP classification

  • OECD FORD branch

    10201 - Computer sciences, information science, bioinformathics (hardware development to be 2.2, social aspect to be 5.8)

Result continuities

  • Project

    <a href="/en/project/GA17-00902S" target="_blank" >GA17-00902S: Advanded Joint Blind Source Separation Methods</a><br>

  • Continuities

    P - Projekt vyzkumu a vyvoje financovany z verejnych zdroju (s odkazem do CEP)

Others

  • Publication year

    2020

  • Confidentiality

    S - Úplné a pravdivé údaje o projektu nepodléhají ochraně podle zvláštních právních předpisů

Data specific for result type

  • Name of the periodical

    IET Signal Processing

  • ISSN

    1751-9675

  • e-ISSN

  • Volume of the periodical

    14

  • Issue of the periodical within the volume

    3

  • Country of publishing house

    GB - UNITED KINGDOM

  • Number of pages

    10

  • Pages from-to

    124-133

  • UT code for WoS article

    000530448800002

  • EID of the result in the Scopus database

    2-s2.0-85084193404