Articles

Autocorrelation-based noise subtraction method with smoothing, overestimation, energy, and cepstral mean and variance normalization for noisy speech recognition

Autocorrelation domain is a proper domain for clean speech signal and noise separation. In this paper, a method is proposed to decrease effects of noise on the clean speech signal, autocorrelation-based noise ...

Authors: Gholamreza Farahani

Two-layer similarity fusion model for cover song identification

Various musical descriptors have been developed for Cover Song Identification (CSI). However, different descriptors are based on various assumptions, designed for representing distinct characteristics of music...

Authors: Ning Chen, Mingyu Li and Haidong Xiao

Robust sound event classification with bilinear multi-column ELM-AE and two-stage ensemble learning

The automatic sound event classification (SEC) has attracted a growing attention in recent years. Feature extraction is a critical factor in SEC system, and the deep neural network (DNN) algorithms have achiev...

Authors: Junjie Zhang, Jie Yin, Qi Zhang, Jun Shi and Yan Li

Effective blind speech watermarking via adaptive mean modulation and package synchronization in DWT domain

This paper outlines a package synchronization scheme for blind speech watermarking in the discrete wavelet transform (DWT) domain. Following two-level DWT decomposition, watermark bits and synchronization code...

Authors: Hwai-Tsu Hu, Shiow-Jyu Lin and Ling-Yuan Hsu

Miscommunication handling in spoken dialog systems based on error-aware dialog state detection

With the exponential growth in computing power and progress in speech recognition technology, spoken dialog systems (SDSs) with which a user interacts through natural speech has been widely used in human-compu...

Authors: Chung-Hsien Wu, Ming-Hsiang Su and Wei-Bin Liang

An MDCT domain three-point interpolation-based low-complexity frequency estimator

Signal frequency estimation is a problem of significance in many applications including audio signal processing. Compressed domain audio frequency estimators that directly use the modified discrete cosine tran...

Authors: Yujie Dun, Guizhong Liu and Xingsong Hou

A computational study of auditory models in music recognition tasks for normal-hearing and hearing-impaired listeners

The benefit of auditory models for solving three music recognition tasks—onset detection, pitch estimation, and instrument recognition—is analyzed. Appropriate features are introduced which enable the use of s...

Authors: Klaus Friedrichs, Nadja Bauer, Rainer Martin and Claus Weihs

Context-dependent factored language models

The incorporation of grammatical information into speech recognition systems is often used to increase performance in morphologically rich languages. However, this introduces demands for sufficiently large tra...

Authors: Gregor Donaj and Zdravko Kačič

Statistical analysis of orthographic and phonemic language corpus for word-based and phoneme-based Polish language modelling

This article presents the original results of Polish language statistical analysis, based on the orthographic and phonemic language corpus. Phonemic language corpus for Polish was developed by using automatic ...

Authors: Piotr Kłosowski

Reversible audio data hiding algorithm using noncausal prediction of alterable orders

This paper presents a reversible data hiding scheme for digital audio by using noncausal prediction of alterable orders. Firstly, the samples in a host signal are divided into the cross and the dot sets. Then, ea...

Authors: Shijun Xiang and Zihao Li

Efficiency of chosen speech descriptors in relation to emotion recognition

This research paper presents parametrization of emotional speech using a pool of common features utilized in emotion recognition such as fundamental frequency, formants, energy, MFCC, PLP, and LPC coefficients. T...

Authors: Dorota Kamińska, Tomasz Sapiński and Gholamreza Anbarjafari

Cantor Digitalis: chironomic parametric synthesis of singing

Cantor Digitalis is a performative singing synthesizer that is composed of two main parts: a chironomic control interface and a parametric voice synthesizer. The control interface is based on a pen/touch graph...

Authors: Lionel Feugère, Christophe d’Alessandro, Boris Doval and Olivier Perrotin

New approach for determining the QoS of MP3-coded voice signals in IP networks

Present-day IP transport platforms being what they are, it will never be possible to rule out conflicts between the available services. The logical consequence of this assertion is the inevitable conclusion th...

Authors: Tadeus Uhl, Stefan Paulsen and Krzysztof Nowicki

Impact of acoustic similarity on efficiency of verbal information transmission via subtle prosodic cues

In this study, we investigate the effect of tiny acoustic differences on the efficiency of prosodic information transmission. Study participants listened to textually ambiguous sentences, which could be unders...

Authors: Bohan Chen, Norihide Kitaoka and Kazuya Takeda

Structure of pauses in speech in the context of speaker verification and classification of speech type

Statistics of pauses appearing in Polish as a potential source of biometry information for automatic speaker recognition were described. The usage of three main types of acoustic pauses (silent, filled and bre...

Authors: Magdalena Igras-Cybulska, Bartosz Ziółko, Piotr Żelasko and Marcin Witkowski

Fast fundamental frequency determination via adaptive autocorrelation

We present an algorithm for the estimation of fundamental frequencies in voiced audio signals. The method is based on an autocorrelation of a signal with a segment of the same signal. During operation, frequen...

Authors: Michael Staudacher, Viktor Steixner, Andreas Griessner and Clemens Zierhofer

Single microphone speech separation by diffusion-based HMM estimation

We present a novel non-iterative and rigorously motivated approach for estimating hidden Markov models (HMMs) and factorial hidden Markov models (FHMMs) of high-dimensional signals. Our approach utilizes the a...

Authors: Yochay R. Yeminy, Yosi Keller and Sharon Gannot

A hybrid input-type recurrent neural network for LVCSR language modeling

Substantial amounts of resources are usually required to robustly develop a language model for an open vocabulary speech recognition system as out-of-vocabulary (OOV) words can hurt recognition accuracy. In th...

Authors: Vataya Chunwijitra, Ananlada Chotimongkol and Chai Wutiwiwatchai

Voice activity detection algorithm based on long-term pitch information

A new voice activity detection algorithm based on long-term pitch divergence is presented. The long-term pitch divergence not only decomposes speech signals with a bionic decomposition but also makes full use ...

Authors: Xu-Kui Yang, Liang He, Dan Qu and Wei-Qiang Zhang

JND-based spatial parameter quantization of multichannel audio signals

In multichannel spatial audio coding (SAC), the accurate representations of virtual sounds and the efficient compressions of spatial parameters are the key to perfect reproduction of spatial sound effects in 3...

Authors: Li Gao, Ruimin Hu, Xiaochen Wang, Gang Li, Yuhong Yang and Weiping Tu

Audio bandwidth extension using ensemble of recurrent neural networks

In audio communication systems, the perceptual audio quality of the reproduced audio signals such as the naturalness of the sound is limited by the available audio bandwidth. In this paper, a wideband to super...

Authors: Xin Liu and Chang-Chun Bao

Novel adaptive muting technique for packet loss concealment of ITU-T G.722 using optimized parametric shaping functions

Adaptive muting method using an optimized parametric shaping function as a part of the ITU-T G.722 Appendix IV packet loss concealment algorithm is proposed. The packet loss concealment algorithm incorporating...

Authors: Bong-Ki Lee and Joon-Hyuk Chang

Wise teachers train better DNN acoustic models

Automatic speech recognition is becoming more ubiquitous as recognition performance improves, capable devices increase in number, and areas of new application open up. Neural network acoustic models that can u...

Authors: Ryan Price, Ken-ichi Iso and Koichi Shinoda

Semi-supervised feature selection for audio classification based on constraint compensated Laplacian score

Audio classification, classifying audio segments into broad categories such as speech, non-speech, and silence, is an important front-end problem in speech signal processing. Dozens of features have been propo...

Authors: Xu-Kui Yang, Liang He, Dan Qu, Wei-Qiang Zhang and Michael T. Johnson

Prosodic mapping of text font based on the dimensional theory of emotions: a case study on style and size

Current text-to-speech systems do not support the effective provision of the semantics and the cognitive aspects of the documents’ typographic cues (e.g., font type, style, and size). A novel approach is intro...

Authors: Dimitrios Tsonos and Georgios Kouroupetroglou

Localization based stereo speech source separation using probabilistic time-frequency masking and deep neural networks

Time-frequency (T-F) masking is an effective method for stereo speech source separation. However, reliable estimation of the T-F mask from sound mixtures is a challenging task, especially when room reverberati...

Authors: Yang Yu, Wenwu Wang and Peng Han

Developing a unit selection voice given audio without corresponding text

Today, a large amount of audio data is available on the web in the form of audiobooks, podcasts, video lectures, video blogs, news bulletins, etc. In addition, we can effortlessly record and store audio data s...

Authors: Tejas Godambe, Sai Krishna Rallabandi, Suryakanth V. Gangashetty, Ashraf Alkhairy and Afshan Jafri

iSargam: music notation representation for Indian Carnatic music

Indian classical music, including its two varieties, Carnatic and Hindustani music, has a rich music tradition and enjoys a wide audience from various parts of the world. The Carnatic music which is more popul...

Authors: Stanly Mammen, Ilango Krishnamurthi, A. Jalaja Varma and G. Sujatha

Hybrid statistical/unit-selection Turkish speech synthesis using suffix units

Unit selection based text-to-speech synthesis (TTS) has been the dominant TTS approach of the last decade. Despite its success, unit selection approach has its disadvantages. One of the most significant disadv...

Authors: Cenk Demiroğlu and Ekrem Güner

Grid-based approximation for voice conversion in low resource environments

The goal of voice conversion is to modify a source speaker’s speech to sound as if spoken by a target speaker. Common conversion methods are based on Gaussian mixture modeling (GMM). They aim to statistically ...

Authors: Hadas Benisty, David Malah and Koby Crammer

Detecting fingering of overblown flute sound using sparse feature learning

In woodwind instruments such as a flute, producing a higher-pitched tone than a standard tone by increasing the blowing pressure is called overblowing, and this allows several distinct fingerings for the same ...

Authors: Yoonchang Han and Kyogu Lee

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