← Search

Lukás Burget

42 accepted papers

2025

CA-MHFA: A Context-Aware Multi-Head Factorized Attentive Pooling for SSL-Based Speaker Verification

ICASSP 2025accepted

Self-supervised learning (SSL) models for speaker verification (SV) have gained significant attention in recent years. However, existing SSL-based SV systems often struggle to capture local temporal dependencies and generalize across different tasks. In this paper, we propose context-aware multi-hea…

Cited by 0SourceScholar
2025

Leveraging Self-Supervised Learning for Speaker Diarization

ICASSP 2025accepted

End-to-end neural diarization has evolved considerably over the past few years, but data scarcity is still a major obstacle for further improvements. Self-supervised learning methods such as WavLM have shown promising performance on several downstream tasks, but their application on speaker diarizat…

Cited by 0SourceScholar
2025

Target Speaker ASR with Whisper

ICASSP 2025accepted

We propose a novel approach to enable the use of large, single-speaker ASR models, such as Whisper, for target speaker ASR. The key claim of this method is that it is much easier to model relative differences among speakers by learning to condition on frame-level diarization outputs than to learn th…

Cited by 0SourceScholar
2025

Text-dependent Speaker Verification Challenge 2024: Exploring Shared and User-defined Passphrases

ICASSP 2025accepted

In contrast to text-independent speaker verification, which has received significant attention from researchers and has many competitions dedicated to it, text-dependent speaker verification (TdSV) has been less explored recently. The TdSV Challenge 2024 was organized to analyze and explore novel me…

Cited by 0SourceScholar
2024

Diacorrect: Error Correction Back-End for Speaker Diarization

ICASSP 2024accepted

In this work, we propose an error correction framework, named DiaCorrect, to refine the output of a diarization system in a simple yet effective way. This method is inspired by error correction techniques in automatic speech recognition. Our model consists of two parallel convolutional encoders and…

Cited by 0SourceScholar
2024

Discriminative Training of VBx Diarization

ICASSP 2024accepted

Bayesian HMM clustering of x-vector sequences (VBx) has become a widely adopted diarization baseline model in publications and challenges. It uses an HMM to model speaker turns, a generatively trained probabilistic linear discriminant analysis (PLDA) for speaker distribution modeling, and Bayesian i…

Cited by 0SourceScholar
2023

Multi-Speaker and Wide-Band Simulated Conversations as Training Data for End-to-End Neural Diarization

ICASSP 2023accepted

End-to-end diarization presents an attractive alternative to standard cascaded diarization systems because a single system can handle all aspects of the task at once. Many flavors of end-to-end models have been proposed but all of them require (so far non-existing) large amounts of annotated data fo…

Cited by 0SourceScholar
2023

Parameter-Efficient Transfer Learning of Pre-Trained Transformer Models for Speaker Verification Using Adapters

ICASSP 2023accepted

Recently, the pre-trained Transformer models have received a rising interest in the field of speech processing thanks to their great success in various downstream tasks. However, most fine-tuning approaches update all the parameters of the pre-trained model, which becomes prohibitive as the model si…

Cited by 0SourceScholar
2023

Speech-Based Emotion Recognition with Self-Supervised Models Using Attentive Channel-Wise Correlations and Label Smoothing

ICASSP 2023accepted

When recognizing emotions from speech, we encounter two common problems: how to optimally capture emotion-relevant information from the speech signal and how to best quantify or categorize the noisy subjective emotion labels. Self-supervised pre-trained representations can robustly capture informati…

Cited by 30SourceScholar
2022

DPCCN: Densely-Connected Pyramid Complex Convolutional Network for Robust Speech Separation and Extraction

ICASSP 2022accepted

In recent years, a number of time-domain speech separation methods have been proposed. However, most of them are very sensitive to the environments and wide domain coverage tasks. In this paper, from the time-frequency domain perspective, we propose a densely-connected pyramid complex convolutional…

Cited by 0SourceScholar
2022

GPU-Accelerated Forward-Backward Algorithm with Application to Lattice-Free MMI

ICASSP 2022accepted

We propose to express the forward-backward algorithm in terms of operations between sparse matrices in a specific semiring. This new perspective naturally leads to a GPU-friendly algorithm which is easy to implement in Julia or any programming languages with native support of semiring algebra. We us…

Cited by 0SourceScholar
2022

Multi-Channel Speaker Verification with Conv-Tasnet Based Beamformer

ICASSP 2022accepted

We focus on the problem of speaker recognition in far-field multichannel data. The main contribution is introducing an alternative way of predicting spatial covariance matrices (SCMs) for a beamformer from the time domain signal. We propose to use ConvTasNet, a well-known source separation model, an…

Cited by 0SourceScholar
2022

Multisv: Dataset for Far-Field Multi-Channel Speaker Verification

ICASSP 2022accepted

Motivated by unconsolidated data situation and the lack of a standard benchmark in the field, we complement our previous efforts and present a comprehensive corpus designed for training and evaluating text-independent multi-channel speaker verification systems. It can be readily used also for experi…

Cited by 0SourceScholar
2021

A Hierarchical Subspace Model for Language-Attuned Acoustic Unit Discovery

ICASSP 2021accepted

In this work, we propose a hierarchical subspace model for acoustic unit discovery. In this approach, we frame the task as one of learning embeddings on a low-dimensional phonetic subspace, and simultaneously specify the subspace itself as an embedding on a hyper-subspace. We train the hyper-subspac…

Cited by 12SourceScholar
2021

Analysis of the but Diarization System for Voxconverse Challenge

ICASSP 2021accepted

This paper describes the system developed by the BUT team for the fourth track of the VoxCeleb Speaker Recognition Challenge, focusing on diarization on the VoxConverse dataset. The system consists of signal pre-processing, voice activity detection, speaker embedding extraction, an initial agglomera…

Cited by 36SourceScholar
2021

Eat: Enhanced ASR-TTS for Self-Supervised Speech Recognition

ICASSP 2021accepted

Self-supervised ASR-TTS models suffer in out-of-domain data conditions. Here we propose an enhanced ASR-TTS (EAT) model that incorporates two main features: 1) The ASR→TTS direction is equipped with a language model reward to penalize the ASR hypotheses before forwarding it to TTS. 2) In the TTS→ASR…

Cited by 0SourceScholar
2021

Jointly Trained Transformers Models for Spoken Language Translation

ICASSP 2021accepted

End-to-End and cascade (ASR-MT) spoken language translation (SLT) systems are reaching comparable performances, however, a large degradation is observed when translating the ASR hypothesis in comparison to using oracle input text. In this work, degradation in performance is reduced by creating an En…

Cited by 0SourceScholar
2020

But System for the Second Dihard Speech Diarization Challenge

ICASSP 2020accepted

This paper describes the winning systems developed by the BUT team for the four tracks of the Second DIHARD Speech Diarization Challenge. For tracks 1 and 2 the systems were mainly based on performing agglomerative hierarchical clustering (AHC) of x-vectors, followed by another x-vector clustering b…

Cited by 60SourceScholar
2020

Investigation of Specaugment for Deep Speaker Embedding Learning

ICASSP 2020accepted

SpecAugment is a newly proposed data augmentation method for speech recognition. By randomly masking bands in the log Mel spectogram this method leads to impressive performance improvements. In this paper, we investigate the usage of SpecAugment for speaker verification tasks. Two different models,…

Cited by 0SourceScholar
2020

Optimizing Bayesian Hmm Based X-Vector Clustering for the Second Dihard Speech Diarization Challenge

ICASSP 2020accepted

This paper presents an analysis of our diarization system winning the second DIHARD speech diarization challenge, track 1. This system is based on clustering x-vector speaker embeddings extracted every 0.25s from short segments of the input recording. In this paper, we focus on the two x-vector clus…

Cited by 0SourceScholar
2019

Discriminatively Re-trained I-vector Extractor for Speaker Recognition

ICASSP 2019accepted

In this work we revisit discriminative training of the i-vector extractor component in the standard speaker verification (SV) system. The motivation of our research lies in the robustness and stability of this large generative model, which we want to preserve, and focus its power towards any intende…

Cited by 5SourceScholar
2019

How to Improve Your Speaker Embeddings Extractor in Generic Toolkits

ICASSP 2019accepted

Recently, speaker embeddings extracted with deep neural networks became the state-of-the-art method for speaker verification. In this paper we aim to facilitate its implementation on a more generic toolkit than Kaldi, which we anticipate to enable further improvements on the method. We examine sever…

Cited by 51SourceScholar
2019

Promising Accurate Prefix Boosting for Sequence-to-sequence ASR

ICASSP 2019accepted

In this paper, we present promising accurate prefix boosting (PAPB), a discriminative training technique for attention based sequence-to-sequence (seq2seq) ASR. PAPB is devised to unify the training and testing scheme effectively. The training procedure involves maximizing the score of each partial…

Cited by 16SourceScholar
2019

Speaker Verification Using End-to-end Adversarial Language Adaptation

ICASSP 2019accepted

In this paper we investigate the use of adversarial domain adaptation for addressing the problem of language mismatch between speaker recognition corpora. In the context of speaker verification, adversarial domain adaptation methods aim at minimizing certain divergences between the distribution that…

Cited by 60SourceScholar
2018

Analysis of Multilingual Blstm Acoustic Model on Low and High Resource Languages

ICASSP 2018accepted

The paper provides an analysis of automatic speech recognition systems (ASR) based on multilingual BLSTM, where we used multi-task training with separate classification layer for each language. The focus is on low resource languages, where only a limited amount of transcribed speech is available. In…

Cited by 0SourceScholar
2018

Bayesian Models for Unit Discovery on a Very Low Resource Language

ICASSP 2018accepted

Developing speech technologies for low-resource languages has become a very active research field over the last decade. Among others, Bayesian models have shown some promising results on artificial examples but still lack of in situ experiments. Our work applies state-of-the-art Bayesian models to u…

Cited by 0SourceScholar
2018

End-to-End DNN Based Speaker Recognition Inspired by I-Vector and PLDA

ICASSP 2018accepted

Recently, several end-to-end speaker verification systems based on deep neural networks (DNNs) have been proposed. These systems have been proven to be competitive for text-dependent tasks as well as for text-independent tasks with short utterances. However, for text-independent tasks with longer ut…

Cited by 56SourceScholar
2018

Out-of-Vocabulary Word Recovery using FST-Based Subword Unit Clustering in a Hybrid ASR System

ICASSP 2018accepted

The paper presents a new approach to extracting useful information from out-of-vocabulary (OOV) speech regions in ASR system output. The system makes use of a hybrid decoding network with both words and sub-word units. In the decoded lattices, candidates for OOV regions are identified as sub-graphs…

Cited by 0SourceScholar
2017

An empirical evaluation of zero resource acoustic unit discovery

ICASSP 2017accepted

Acoustic unit discovery (AUD) is a process of automatically identifying a categorical acoustic unit inventory from speech and producing corresponding acoustic unit tokenizations. AUD provides an important avenue for unsupervised acoustic model training in a zero resource setting where expert-provide…

Cited by 0SourceScholar
2017

Bayesian joint-sequence models for grapheme-to-phoneme conversion

ICASSP 2017accepted

We describe a fully Bayesian approach to grapheme-to-phoneme conversion based on the joint-sequence model (JSM). Usually, standard smoothed n-gram language models (LM, e.g. Kneser-Ney) are used with JSMs to model graphone sequences (joint grapheme-phoneme pairs). However, we take a Bayesian approach…

Cited by 0SourceScholar
2017

Bayesian phonotactic Language Model for Acoustic Unit Discovery

ICASSP 2017accepted

Recent work on Acoustic Unit Discovery (AUD) has led to the development of a non-parametric Bayesian phone-loop model where the prior over the probability of the phone-like units is assumed to be sampled from a Dirichlet Process (DP). In this work, we propose to improve this model by incorporating a…

Cited by 0SourceScholar
2017

Residual memory networks: Feed-forward approach to learn long-term temporal dependencies

ICASSP 2017accepted

Training deep recurrent neural network (RNN) architectures is complicated due to the increased network complexity. This disrupts the learning of higher order abstracts using deep RNN. In case of feed-forward networks training deep structures is simple and faster while learning long-term temporal inf…

Cited by 0SourceScholar
2017

Topic identification of spoken documents using unsupervised acoustic unit discovery

ICASSP 2017accepted

This paper investigates the application of unsupervised acoustic unit discovery for topic identification (topic ID) of spoken audio documents. The acoustic unit discovery method is based on a non-parametric Bayesian phone-loop model that segments a speech utterance into phone-like categories. The di…

Cited by 0SourceScholar
2016

Analysis of DNN approaches to speaker identification

ICASSP 2016accepted

This work studies the usage of the Deep Neural Network (DNN) Bottleneck (BN) features together with the traditional MFCC features in the task of i-vector-based speaker recognition. We decouple the sufficient statistics extraction by using separate GMM models for frame alignment, and for statistics n…

Cited by 0SourceScholar
2016

Audio enhancing with DNN autoencoder for speaker recognition

ICASSP 2016accepted

In this paper we present a design of a DNN-based autoencoder for speech enhancement and its use for speaker recognition systems for distant microphones and noisy data. We started with augmenting the Fisher database with artificially noised and reverberated data and trained the autoencoder to map noi…

Cited by 0SourceScholar
2016

Multilingual region-dependent transforms

ICASSP 2016accepted

In recent years, trained feature extraction (FE) schemes based on neural networks have replaced or complemented traditional approaches in top performing systems. This paper deals with FE in multilingual scenarios with a target language with low amount of transcribed data. Continuing our previous wor…

Cited by 0SourceScholar
2016

Sequence summarizing neural network for speaker adaptation

ICASSP 2016accepted

In this paper, we propose a DNN adaptation technique, where the i-vector extractor is replaced by a Sequence Summarizing Neural Network (SSNN). Similarly to i-vector extractor, the SSNN produces a "summary vector", representing an acoustic summary of an utterance. Such vector is then appended to the…

Cited by 0SourceScholar
2015

Copingwith channel mismatch in Query-by-Example - But QUESST 2014

ICASSP 2015accepted

The paper investigates into Query by Example (QbE) - a spoken term detection technique with queries entered by voice. It describes BUT QbE system that achieved the best accuracy in MediaEval QUESST2014 evaluations. This evaluation was challenging because of severe mismatch between queries and uttera…

Cited by 0SourceScholar
2015

Employment of Subspace Gaussian Mixture Models in speaker recognition

ICASSP 2015accepted

This paper presents Subspace Gaussian Mixture Model (SGMM) approach employed as a probabilistic generative model to estimate speaker vector representations to be subsequently used in the speaker verification task. SGMMs have already been shown to significantly outperform traditional HMM/GMMs in Auto…

Cited by 29SourceScholar
2015

Towards machines that know when they do not know: Summary of work done at 2014 Frederick Jelinek Memorial Workshop

ICASSP 2015accepted

A group of junior and senior researchers gathered as a part of the 2014 Frederick Jelinek Memorial Workshop in Prague to address the problem of predicting the accuracy of a nonlinear Deep Neural Network probability estimator for unknown data in a different application domain from the domain in which…

Cited by 0SourceScholar