Search Results for author: Tomohiro Nakatani

Found 29 papers, 5 papers with code

Improving speaker discrimination of target speech extraction with time-domain SpeakerBeam

1 code implementation23 Jan 2020 Marc Delcroix, Tsubasa Ochiai, Katerina Zmolikova, Keisuke Kinoshita, Naohiro Tawara, Tomohiro Nakatani, Shoko Araki

First, we propose a time-domain implementation of SpeakerBeam similar to that proposed for a time-domain audio separation network (TasNet), which has achieved state-of-the-art performance for speech separation.

Speaker Identification Speech Extraction

PILOT: Introducing Transformers for Probabilistic Sound Event Localization

1 code implementation7 Jun 2021 Christopher Schymura, Benedikt Bönninghoff, Tsubasa Ochiai, Marc Delcroix, Keisuke Kinoshita, Tomohiro Nakatani, Shoko Araki, Dorothea Kolossa

Sound event localization aims at estimating the positions of sound sources in the environment with respect to an acoustic receiver (e. g. a microphone array).

Event Detection

Integration of variational autoencoder and spatial clustering for adaptive multi-channel neural speech separation

1 code implementation24 Nov 2020 Katerina Zmolikova, Marc Delcroix, Lukáš Burget, Tomohiro Nakatani, Jan "Honza" Černocký

In this paper, we propose a method combining variational autoencoder model of speech with a spatial clustering approach for multi-channel speech separation.

Audio and Speech Processing

Exploiting Attention-based Sequence-to-Sequence Architectures for Sound Event Localization

1 code implementation28 Feb 2021 Christopher Schymura, Tsubasa Ochiai, Marc Delcroix, Keisuke Kinoshita, Tomohiro Nakatani, Shoko Araki, Dorothea Kolossa

Herein, attentions allow for capturing temporal dependencies in the audio signal by focusing on specific frames that are relevant for estimating the activity and direction-of-arrival of sound events at the current time-step.

Automatic Speech Recognition Automatic Speech Recognition (ASR) +1

All-neural online source separation, counting, and diarization for meeting analysis

no code implementations21 Feb 2019 Thilo von Neumann, Keisuke Kinoshita, Marc Delcroix, Shoko Araki, Tomohiro Nakatani, Reinhold Haeb-Umbach

While significant progress has been made on individual tasks, this paper presents for the first time an all-neural approach to simultaneous speaker counting, diarization and source separation.

Automatic Speech Recognition Automatic Speech Recognition (ASR) +3

Jointly optimal dereverberation and beamforming

no code implementations30 Oct 2019 Christoph Boeddeker, Tomohiro Nakatani, Keisuke Kinoshita, Reinhold Haeb-Umbach

We previously proposed an optimal (in the maximum likelihood sense) convolutional beamformer that can perform simultaneous denoising and dereverberation, and showed its superiority over the widely used cascade of a WPE dereverberation filter and a conventional MPDR beamformer.

Denoising

End-to-end training of time domain audio separation and recognition

no code implementations18 Dec 2019 Thilo von Neumann, Keisuke Kinoshita, Lukas Drude, Christoph Boeddeker, Marc Delcroix, Tomohiro Nakatani, Reinhold Haeb-Umbach

The rising interest in single-channel multi-speaker speech separation sparked development of End-to-End (E2E) approaches to multi-speaker speech recognition.

Speaker Recognition speech-recognition +2

Tackling real noisy reverberant meetings with all-neural source separation, counting, and diarization system

no code implementations9 Mar 2020 Keisuke Kinoshita, Marc Delcroix, Shoko Araki, Tomohiro Nakatani

Automatic meeting analysis is an essential fundamental technology required to let, e. g. smart devices follow and respond to our conversations.

speaker-diarization Speaker Diarization +1

Multi-talker ASR for an unknown number of sources: Joint training of source counting, separation and ASR

no code implementations4 Jun 2020 Thilo von Neumann, Christoph Boeddeker, Lukas Drude, Keisuke Kinoshita, Marc Delcroix, Tomohiro Nakatani, Reinhold Haeb-Umbach

Most approaches to multi-talker overlapped speech separation and recognition assume that the number of simultaneously active speakers is given, but in realistic situations, it is typically unknown.

Automatic Speech Recognition Automatic Speech Recognition (ASR) +2

Block Coordinate Descent Algorithms for Auxiliary-Function-Based Independent Vector Extraction

no code implementations18 Oct 2020 Rintaro Ikeshita, Tomohiro Nakatani, Shoko Araki

We also newly develop a BCD for a semiblind IVE in which the transfer functions for several super-Gaussian sources are given a priori.

Neural Network-based Virtual Microphone Estimator

no code implementations12 Jan 2021 Tsubasa Ochiai, Marc Delcroix, Tomohiro Nakatani, Rintaro Ikeshita, Keisuke Kinoshita, Shoko Araki

Developing microphone array technologies for a small number of microphones is important due to the constraints of many devices.

Speech Enhancement

Speaker activity driven neural speech extraction

no code implementations14 Jan 2021 Marc Delcroix, Katerina Zmolikova, Tsubasa Ochiai, Keisuke Kinoshita, Tomohiro Nakatani

Target speech extraction, which extracts the speech of a target speaker in a mixture given auxiliary speaker clues, has recently received increased interest.

Speech Extraction

Multimodal Attention Fusion for Target Speaker Extraction

no code implementations2 Feb 2021 Hiroshi Sato, Tsubasa Ochiai, Keisuke Kinoshita, Marc Delcroix, Tomohiro Nakatani, Shoko Araki

Recently an audio-visual target speaker extraction has been proposed that extracts target speech by using complementary audio and visual clues.

Target Speaker Extraction

Independent Vector Extraction for Fast Joint Blind Source Separation and Dereverberation

no code implementations9 Feb 2021 Rintaro Ikeshita, Tomohiro Nakatani

We address a blind source separation (BSS) problem in a noisy reverberant environment in which the number of microphones $M$ is greater than the number of sources of interest, and the other noise components can be approximated as stationary and Gaussian distributed.

blind source separation

Comparison of remote experiments using crowdsourcing and laboratory experiments on speech intelligibility

no code implementations17 Apr 2021 Ayako Yamamoto, Toshio Irino, Kenichi Arai, Shoko Araki, Atsunori Ogawa, Keisuke Kinoshita, Tomohiro Nakatani

Many subjective experiments have been performed to develop objective speech intelligibility measures, but the novel coronavirus outbreak has made it very difficult to conduct experiments in a laboratory.

Speech Enhancement

ISS2: An Extension of Iterative Source Steering Algorithm for Majorization-Minimization-Based Independent Vector Analysis

no code implementations2 Feb 2022 Rintaro Ikeshita, Tomohiro Nakatani

Although the time complexity per iteration of ISS is $m$ times smaller than that of IP, the conventional ISS converges slower than the current fastest IP (called $\text{IP}_2$) that updates two rows of $W$ in each iteration.

Listen only to me! How well can target speech extraction handle false alarms?

no code implementations11 Apr 2022 Marc Delcroix, Keisuke Kinoshita, Tsubasa Ochiai, Katerina Zmolikova, Hiroshi Sato, Tomohiro Nakatani

Target speech extraction (TSE) extracts the speech of a target speaker in a mixture given auxiliary clues characterizing the speaker, such as an enrollment utterance.

Speaker Identification Speaker Verification +2

Mask-based Neural Beamforming for Moving Speakers with Self-Attention-based Tracking

no code implementations7 May 2022 Tsubasa Ochiai, Marc Delcroix, Tomohiro Nakatani, Shoko Araki

We thus introduce a learning-based framework that computes optimal attention weights for beamforming.

Modified Parametric Multichannel Wiener Filter \\for Low-latency Enhancement of Speech Mixtures with Unknown Number of Speakers

no code implementations29 Jun 2023 Ning Guo, Tomohiro Nakatani, Shoko Araki, Takehiro Moriya

This paper introduces a novel low-latency online beamforming (BF) algorithm, named Modified Parametric Multichannel Wiener Filter (Mod-PMWF), for enhancing speech mixtures with unknown and varying number of speakers.

Low-latency processing

Target Speech Extraction with Conditional Diffusion Model

no code implementations8 Aug 2023 Naoyuki Kamo, Marc Delcroix, Tomohiro Nakatani

TSE is realized by conditioning the extraction process on a clue identifying the target speaker.

Denoising Speech Denoising +2

Array Geometry-Robust Attention-Based Neural Beamformer for Moving Speakers

no code implementations5 Feb 2024 Marvin Tammen, Tsubasa Ochiai, Marc Delcroix, Tomohiro Nakatani, Shoko Araki, Simon Doclo

Recently, a mask-based beamformer with attention-based spatial covariance matrix aggregator (ASA) was proposed, which was demonstrated to track moving sources accurately.

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