Articles

Comparison of ALBAYZIN query-by-example spoken term detection 2012 and 2014 evaluations

Query-by-example spoken term detection (QbE STD) aims at retrieving data from a speech repository given an acoustic query containing the term of interest as input. Nowadays, it is receiving much interest due t...

Authors: Javier Tejedor, Doroteo T. Toledano, Paula Lopez-Otero, Laura Docio-Fernandez and Carmen Garcia-Mateo

Speech signal modeling using multivariate distributions

Using a proper distribution function for speech signal or for its representations is of crucial importance in statistical-based speech processing algorithms. Although the most commonly used probability density...

Authors: Ali Aroudi, Hadi Veisi, Hossein Sameti and Zahra Mafakheri

A multichannel diffuse power estimator for dereverberation in the presence of multiple sources

Using a recently proposed informed spatial filter, it is possible to effectively and robustly reduce reverberation from speech signals captured in noisy environments using multiple microphones. Late reverberat...

Authors: Sebastian Braun and Emanuël A. P. Habets

Albayzín-2014 evaluation: audio segmentation and classification in broadcast news domains

Audio segmentation is important as a pre-processing task to improve the performance of many speech technology tasks and, therefore, it has an undoubted research interest. This paper describes the database, the...

Authors: Diego Castán, David Tavarez, Paula Lopez-Otero, Javier Franco-Pedroso, Héctor Delgado, Eva Navas, Laura Docio-Fernández, Daniel Ramos, Javier Serrano, Alfonso Ortega and Eduardo Lleida

Small-parallel exemplar-based voice conversion in noisy environments using affine non-negative matrix factorization

The need to have a large amount of parallel data is a large hurdle for the practical use of voice conversion (VC). This paper presents a novel framework of exemplar-based VC that only requires a small number o...

Authors: Ryo Aihara, Takao Fujii, Toru Nakashika, Tetsuya Takiguchi and Yasuo Ariki

Semi-fragile digital speech watermarking for online speaker recognition

In this paper, a semi-fragile and blind digital speech watermarking technique for online speaker recognition systems based on the discrete wavelet packet transform (DWPT) and quantization index modulation (QIM...

Authors: Mohammad Ali Nematollahi, Mohammad Ali Akhaee, S. A. R. Al-Haddad and Hamurabi Gamboa-Rosales

Erratum to: Efficient voice activity detection algorithm using long-term spectral flatness measure

Authors: Yanna Ma and Akinori Nishihara

Physical task stress and speaker variability in voice quality

The presence of physical task stress induces changes in the speech production system which in turn produces changes in speaking behavior. This results in measurable acoustic correlates including changes to for...

Authors: Keith W. Godin and John H. L. Hansen

Speech enhancement based on Bayesian decision and spectral amplitude estimation

In this paper, a single-channel speech enhancement method based on Bayesian decision and spectral amplitude estimation is proposed, in which the speech detection module and spectral amplitude estimation module...

Authors: Feng Deng and Chang-Chun Bao

Biomimetic spectro-temporal features for music instrument recognition in isolated notes and solo phrases

The identity of musical instruments is reflected in the acoustic attributes of musical notes played with them. Recently, it has been argued that these characteristics of musical identity (or timbre) can be bes...

Authors: Kailash Patil and Mounya Elhilali

Exploiting spectro-temporal locality in deep learning based acoustic event detection

In recent years, deep learning has not only permeated the computer vision and speech recognition research fields but also fields such as acoustic event detection (AED). One of the aims of AED is to detect and ...

Authors: Miquel Espi, Masakiyo Fujimoto, Keisuke Kinoshita and Tomohiro Nakatani

Phone recognition with hierarchical convolutional deep maxout networks

Deep convolutional neural networks (CNNs) have recently been shown to outperform fully connected deep neural networks (DNNs) both on low-resource and on large-scale speech tasks. Experiments indicate that conv...

Authors: László Tóth

Multimodal voice conversion based on non-negative matrix factorization

A multimodal voice conversion (VC) method for noisy environments is proposed. In our previous non-negative matrix factorization (NMF)-based VC method, source and target exemplars are extracted from parallel tr...

Authors: Kenta Masaka, Ryo Aihara, Tetsuya Takiguchi and Yasuo Ariki

The Latin Music Mood Database

In this paper we present the Latin Music Mood Database, an extension of the Latin Music Database but for the task of music mood/emotion classification. The method for assigning mood labels to the musical recor...

Authors: Carolina L. dos Santos and Carlos N. Silla Jr

Regularized minimum class variance extreme learning machine for language recognition

Support vector machines (SVMs) have played an important role in the state-of-the-art language recognition systems. The recently developed extreme learning machine (ELM) tends to have better scalability and ach...

Authors: Jiaming Xu, Wei-Qiang Zhang, Jia Liu and Shanhong Xia

Spoken term detection ALBAYZIN 2014 evaluation: overview, systems, results, and discussion

Spoken term detection (STD) aims at retrieving data from a speech repository given a textual representation of the search term. Nowadays, it is receiving much interest due to the large volume of multimedia inf...

Authors: Javier Tejedor, Doroteo T. Toledano, Paula Lopez-Otero, Laura Docio-Fernandez, Carmen Garcia-Mateo, Antonio Cardenal, Julian David Echeverry-Correa, Alejandro Coucheiro-Limeres, Julia Olcoz and Antonio Miguel

Advanced acoustic modelling techniques in MP3 speech recognition

The automatic recognition of MP3 compressed speech presents a challenge to the current systems due to the lossy nature of compression which causes irreversible degradation of the speech wave. This article eval...

Authors: Michal Borsky, Petr Pollak and Petr Mizera

Emotion in the singing voice—a deeperlook at acoustic features in the light ofautomatic classification

We investigate the automatic recognition of emotions in the singing voice and study the worth and role of a variety of relevant acoustic parameters. The data set contains phrases and vocalises sung by eight re...

Authors: Florian Eyben, Gláucia L Salomão, Johan Sundberg, Klaus R Scherer and Björn W Schuller

An improved i-vector extraction algorithm for speaker verification

Over recent years, i-vector-based framework has been proven to provide state-of-the-art performance in speaker verification. Each utterance is projected onto a total factor space and is represented by a low-di...

Authors: Wei Li, Tianfan Fu and Jie Zhu

Exploiting foreign resources for DNN-based ASR

Manual transcription of audio databases for the development of automatic speech recognition (ASR) systems is a costly and time-consuming process. In the context of deriving acoustic models adapted to a specifi...

Authors: Petr Motlicek, David Imseng, Blaise Potard, Philip N. Garner and Ivan Himawan

Singer identification using perceptual features and cepstral coefficients of an audio signal from Indian video songs

Singer identification is a difficult topic in music information retrieval because background instrumental music is included with singing voice which reduces performance of a system. One of the main disadvantag...

Authors: Tushar Ratanpara and Narendra Patel

Stereo-based histogram equalization for robust speech recognition

Optimal automatic speech recognition (ASR) takes place when the recognition system is tested under circumstances identical to those in which it was trained. However, in the actual real world, there exist many ...

Authors: Randa Al-Wakeel, Mahmoud Shoman, Magdy Aboul-Ela and Sherif Abdou

Robust design of Farrow-structure-based steerable broadband beamformers with sparse tap weights via convex optimization

The Farrow-structure-based steerable broadband beamformer (FSBB) is particularly useful in the applications where sound source of interest may move around a wide angular range. However, in contrast with conven...

Authors: Tiannan Wang and Huawei Chen

ViSQOL: an objective speech quality model

This paper presents an objective speech quality model, ViSQOL, the Virtual Speech Quality Objective Listener. It is a signal-based, full-reference, intrusive metric that models human speech quality perception ...

Authors: Andrew Hines, Jan Skoglund, Anil C Kokaram and Naomi Harte

Deep neural network-based bottleneck feature and denoising autoencoder-based dereverberation for distant-talking speaker identification

Deep neural network (DNN)-based approaches have been shown to be effective in many automatic speech recognition systems. However, few works have focused on DNNs for distant-talking speaker recognition. In this...

Authors: Zhaofeng Zhang, Longbiao Wang, Atsuhiko Kai, Takanori Yamada, Weifeng Li and Masahiro Iwahashi

Lightweight multi-DOA tracking of mobile speech sources

Estimating the directions of arrival (DOAs) of multiple simultaneous mobile sound sources is an important step for various audio signal processing applications. In this contribution, we present an approach tha...

Authors: Caleb Rascon, Gibran Fuentes and Ivan Meza

An acoustic data transmission system based on audio data hiding: method and performance evaluation

Acoustic data transmission (ADT) forms a branch of the audio data hiding techniques with its capability of communicating data in short-range aerial space between a loudspeaker and a microphone. In this paper, ...

Authors: Kiho Cho, Jae Choi and Nam Soo Kim

Evaluation of linguistic and prosodic features for detection of Alzheimer’s disease in Turkish conversational speech

Automatic diagnosis and monitoring of Alzheimer’s disease can have a significant impact on society as well as the well-being of patients. The part of the brain cortex that processes language abilities is one o...

Authors: Ali Khodabakhsh, Fatih Yesil, Ekrem Guner and Cenk Demiroglu

Voice conversion using speaker-dependent conditional restricted Boltzmann machine

This paper presents a voice conversion (VC) method that utilizes conditional restricted Boltzmann machines (CRBMs) for each speaker to obtain high-order speaker-independent spaces where voice features are conv...

Authors: Toru Nakashika, Tetsuya Takiguchi and Yasuo Ariki

An investigation of supervector regression for forensic voice comparison on small data

Automatic forensic voice comparison (FVC) systems employed in forensic casework have often relied on Gaussian Mixture Model - Universal Background Models (GMM-UBMs) for modelling with relatively little researc...

Authors: Chee Cheun Huang, Julien Epps and Tharmarajah Thiruvaran

SIFT-based local spectrogram image descriptor: a novel feature for robust music identification

Music identification via audio fingerprinting has been an active research field in recent years. In the real-world environment, music queries are often deformed by various interferences which typically include...

Authors: Xiu Zhang, Bilei Zhu, Linwei Li, Wei Li, Xiaoqiang Li, Wei Wang, Peizhong Lu and Wenqiang Zhang

A signal subspace approach to spatio-temporal prediction for multichannel speech enhancement

The spatio-temporal-prediction (STP) method for multichannel speech enhancement has recently been proposed. This approach makes it theoretically possible to attenuate the residual noise without distorting spee...

Authors: Adam Borowicz

Within and cross-corpus speech emotion recognition using latent topic model-based features

Owing to the suprasegmental behavior of emotional speech, turn-level features have demonstrated a better success than frame-level features for recognition-related tasks. Conventionally, such features are obtai...

Authors: Mohit Shah, Chaitali Chakrabarti and Andreas Spanias

A novel hybrid of genetic algorithm and ANN for developing a high efficient method for vocal fold pathology diagnosis

In this paper, an initial feature vector based on the combination of the wavelet packet decomposition (WPD) and the Mel frequency cepstral coefficients (MFCCs) is proposed. For optimizing the initial feature v...

Authors: Vahid Majidnezhad

Noisy training for deep neural networks in speech recognition

Deep neural networks (DNNs) have gained remarkable success in speech recognition, partially attributed to the flexibility of DNN models in learning complex patterns of speech signals. This flexibility, however...

Authors: Shi Yin, Chao Liu, Zhiyong Zhang, Yiye Lin, Dong Wang, Javier Tejedor, Thomas Fang Zheng and Yinguo Li

Simulation of tremulous voices using a biomechanical model

Vocal tremor has been simulated using a high-dimensional discrete vocal fold model. Specifically, respiratory, phonatory, and articulatory tremors have been modeled as instabilities in six parameters of the mo...

Authors: Rubén Fraile, Juan Ignacio Godino-Llorente and Malte Kob

Homogenous ensemble phonotactic language recognition based on SVM supervector reconstruction

Currently, acoustic spoken language recognition (SLR) and phonotactic SLR systems are widely used language recognition systems. To achieve better performance, researchers combine multiple subsystems with the r...

Authors: Wei-Wei Liu, Wei-Qiang Zhang, Michael T Johnson and Jia Liu

The self-taught vocal interface

Speech technology is firmly rooted in daily life, most notably in command-and-control (C&C) applications. C&C usability downgrades quickly, however, when used by people with non-standard speech. We pursue a fu...

Authors: Bart Ons, Jort F Gemmeke and Hugo Van hamme

Audio bandwidth extension based on temporal smoothing cepstral coefficients

In this paper, we propose a wideband (WB) to super-wideband audio bandwidth extension (BWE) method based on temporal smoothing cepstral coefficients (TSCC). A temporal relationship of audio signals is included...

Authors: Xin Liu and Chang-Chun Bao

A sub-band-based feature reconstruction approach for robust speaker recognition

Although the field of automatic speaker or speech recognition has been extensively studied over the past decades, the lack of robustness has remained a major challenge. The missing data technique (MDT) is a pr...

Authors: Furong Yan, Yanbin Zhang and Jiachang Yan

Sparse coding of the modulation spectrum for noise-robust automatic speech recognition

The full modulation spectrum is a high-dimensional representation of one-dimensional audio signals. Most previous research in automatic speech recognition converted this very rich representation into the equiv...

Authors: Sara Ahmadi, Seyed Mohammad Ahadi, Bert Cranen and Lou Boves

A cross-lingual adaptation approach for rapid development of speech recognizers for learning disabled users

Building a voice-operated system for learning disabled users is a difficult task that requires a considerable amount of time and effort. Due to the wide spectrum of disabilities and their different related pho...

Authors: Marek Bohac, Michaela Kucharova, Zoraida Callejas, Jan Nouza and Petr Červa

A uniform phase representation for the harmonic model in speech synthesis applications

Feature-based vocoders, e.g., STRAIGHT, offer a way to manipulate the perceived characteristics of the speech signal in speech transformation and synthesis. For the harmonic model, which provide excellent perc...

Authors: Gilles Degottex and Daniel Erro

An imperceptible and robust audio watermarking algorithm

In this paper, we propose a semi-blind, imperceptible, and robust digital audio watermarking algorithm. The proposed algorithm is based on cascading two well-known transforms: the discrete wavelet transform an...

Authors: Ali Al-Haj

Robust Bayesian estimation for context-based speech enhancement

Model-based speech enhancement algorithms that employ trained models, such as codebooks, hidden Markov models, Gaussian mixture models, etc., containing representations of speech such as linear predictive coef...

Authors: Devireddy Hanumantha Rao Naidu and Sriram Srinivasan

The Ethnic Lyrics Fetcher tool

The task of automatic retrieval and extraction of lyrics from the web is of great importance to different Music Information Retrieval applications. However, despite its importance, very little research has bee...

Authors: Rafael P Ribeiro, Murilo AP Almeida and Carlos N Silla Jr

Audio segmentation-by-classification approach based on factor analysis in broadcast news domain

This paper studies a novel audio segmentation-by-classification approach based on factor analysis. The proposed technique compensates the within-class variability by using class-dependent factor loading matric...

Authors: Diego Castán, Alfonso Ortega, Antonio Miguel and Eduardo Lleida

A modified Wiener filtering method combined with wavelet thresholding multitaper spectrum for speech enhancement

This paper proposes a new speech enhancement (SE) algorithm utilizing constraints to the Wiener gain function which is capable of working at 10 dB and lower signal-to-noise ratios (SNRs). The wavelet threshold...

Authors: Yanna Ma and Akinori Nishihara

The influence of speech rate on Fujisaki model parameters

The current paper examines influences of speech rate on Fujisaki model parameters based on read speech from the BonnTempo-Corpus containing productions by 12 native speakers of German at five different intende...

Authors: Hansjörg Mixdorff, Adrian Leemann and Volker Dellwo

Joint estimation of pitch and direction of arrival: improving robustness and accuracy for multi-speaker scenarios

In many speech communication applications, robust localization and tracking of multiple speakers in noisy and reverberant environments are of major importance. Several algorithms to tackle this problem have be...

Authors: Stephan Gerlach, Jörg Bitzer, Stefan Goetze and Simon Doclo