Articles

Semantic structures of timbre emerging from social and acoustic descriptions of music

The perceptual attributes of timbre have inspired a considerable amount of multidisciplinary research, but because of the complexity of the phenomena, the approach has traditionally been confined to laboratory...

Authors: Rafael Ferrer and Tuomas Eerola

System for fast lexical and phonetic spoken term detection in a Czech cultural heritage archive

The main objective of the work presented in this paper was to develop a complete system that would accomplish the original visions of the MALACH project. Those goals were to employ automatic speech recognition...

Authors: Josef Psutka, Jan Švec, Josef V Psutka, Jan Vaněk, Aleš Pražák, Luboš Šmídl and Pavel Ircing

Noise-robust speech feature processing with empirical mode decomposition

In this article, a novel technique based on the empirical mode decomposition methodology for processing speech features is proposed and investigated. The empirical mode decomposition generalizes the Fourier an...

Authors: Kuo-Hau Wu, Chia-Ping Chen and Bing-Feng Yeh

Correlation analysis of the speech multiscale product for the open quotient estimation

This article proposes a multiscale product (MP)-based method for estimating the open quotient (OQ) from the speech waveform. The MP is operated by calculating the wavelet transform coefficients of the speech s...

Authors: Wafa Saidi, Aicha Bouzid and Noureddine Ellouze

An improved adaptive gain equalizer for noise reduction with low speech distortion

In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ra...

Authors: Markus Borgh, Magnus Berggren, Christian Schüldt, Fredric Lindström and Ingvar Claesson

A large vocabulary continuous speech recognition system for Persian language

The first large vocabulary speech recognition system for the Persian language is introduced in this paper. This continuous speech recognition system uses most standard and state-of-the-art speech and language ...

Authors: Hossein Sameti, Hadi Veisi, Mohammad Bahrani, Bagher Babaali and Khosro Hosseinzadeh

Noise reduction for periodic signals using high-resolution frequency analysis

The spectrum subtraction method is one of the most common methods by which to remove noise from a spectrum. Like many noise reduction methods, the spectrum subtraction method uses discrete Fourier transform (D...

Authors: Toshio Yoshizawa, Shigeki Hirobayashi and Tadanobu Misawa

Multi-label classification of music by emotion

This work studies the task of automatic emotion detection in music. Music may evoke more than one different emotion at the same time. Single-label classification and regression cannot model this multiplicity. ...

Authors: Konstantinos Trohidis, Grigorios Tsoumakas, George Kalliris and Ioannis Vlahavas

Robust time delay estimation for speech signals using information theory: A comparison study

Time delay estimation (TDE) is a fundamental subsystem for a speaker localization and tracking system. Most of the traditional TDE methods are based on second-order statistics (SOS) under Gaussian assumption f...

Authors: Fei Wen and Qun Wan

Semitone frequency mapping to improve music representation for nucleus cochlear implants

The frequency-to-channel mapping for Cochlear implant (CI) signal processors was originally designed to optimize speech perception and generally does not preserve the harmonic structure of music sounds. An alg...

Authors: Sherif Abdellatif Omran, Waikong Lai, Michael Büchler and Norbert Dillier

Audio segmentation of broadcast news in the Albayzin-2010 evaluation: overview, results, and discussion

Recently, audio segmentation has attracted research interest because of its usefulness in several applications like audio indexing and retrieval, subtitling, monitoring of acoustic scenes, etc. Moreover, a pre...

Authors: Taras Butko and Climent Nadeu

Scalable Audio-Content Analysis

Authors: Bhiksha Raj, Paris Smaragdis, Malcolm Slaney, Chung-Hsien Wu, Liming Chen and Hyoung-Gook Kim

Environmental Sound Synthesis, Processing, and Retrieval

Authors: Andrea Valle

Phoneme and Sentence-Level Ensembles for Speech Recognition

We address the question of whether and how boosting and bagging can be used for speech recognition. In order to do this, we compare two different boosting schemes, one at the phoneme level and one at the utter...

Authors: Christos Dimitrakakis and Samy Bengio

Pitch Ranking, Melody Contour and Instrument Recognition Tests Using Two Semitone Frequency Maps for Nucleus Cochlear Implants

To overcome harmonic structure distortions of complex tones in the low frequency range due to the frequency to electrode mapping function used in Nucleus cochlear implants, two modified frequency maps based on...

Authors: Sherif A. Omran, Waikong Lai and Norbert Dillier

Physically Motivated Environmental Sound Synthesis for Virtual Worlds

A system is described for simulating environmental sound in interactive virtual worlds, using the physical state of objects as control parameters. It contains a unified framework for integration with physics s...

Authors: Dylan Menzies

Multiple Source Localization Based on Acoustic Map De-Emphasis

This paper describes a novel approach for localization of multiple sources overlapping in time. The proposed algorithm relies on acoustic maps computed in multi-microphone settings, which are descriptions of t...

Authors: Alessio Brutti, Maurizio Omologo and Piergiorgio Svaizer

Monaural Voiced Speech Segregation Based on Dynamic Harmonic Function

Correlogram is an important representation for periodic signals. It is widely used in pitch estimation and source separation. For these applications, major problems of correlogram are its low resolution and re...

Authors: Xueliang Zhang, Wenju Liu and Bo Xu

Ecological Acoustics Perspective for Content-Based Retrieval of Environmental Sounds

In this paper we present a method to search for environmental sounds in large unstructured databases of user-submitted audio, using a general sound events taxonomy from ecological acoustics. We discuss the use...

Authors: Gerard Roma, Jordi Janer, Stefan Kersten, Mattia Schirosa, Perfecto Herrera and Xavier Serra

A Novel MPEG Audio Degrouping Algorithm and Its Architecture Design

Degrouping is the key component in MPEG Layer II audio decoding. It mainly contains the arithmetic operations of division and modulo. So far no dedicated degrouping algorithm and architecture is well realized....

Authors: Tsung-Han Tsai

An Ontological Framework for Retrieving Environmental Sounds Using Semantics and Acoustic Content

Organizing a database of user-contributed environmental sound recordings allows sound files to be linked not only by the semantic tags and labels applied to them, but also to other sounds with similar acoustic...

Authors: Gordon Wichern, Brandon Mechtley, Alex Fink, Harvey Thornburg and Andreas Spanias

The Effect of a Voice Activity Detector on the Speech Enhancement Performance of the Binaural Multichannel Wiener Filter

A multimicrophone speech enhancement algorithm for binaural hearing aids that preserves interaural time delays was proposed recently. The algorithm is based on multichannel Wiener filtering and relies on a voi...

Authors: Jasmina Catic, Torsten Dau, JörgM Buchholz and Fredrik Gran

Evaluating Environmental Sounds from a Presence Perspective for Virtual Reality Applications

We propose a methodology to design and evaluate environmental sounds for virtual environments. We propose to combine physically modeled sound events with recorded soundscapes. Physical models are used to provi...

Authors: Rolf Nordahl

Instrumental Estimation of E-Model Parameters for Wideband Speech Codecs

A method is described for quantifying the quality of wideband speech codecs. Two parameters are derived from signal-based speech quality model estimations: (i) a wideband equipment impairment factor

Authors: Sebastian Möller, Nicolas Côté, Valérie Gautier-Turbin, Nobuhiko Kitawaki and Akira Takahashi

Optimizing the Directivity of Multiway Loudspeaker Systems

In multiway loudspeaker systems, digital signal processing techniques have been used to correct the frequency response, the propagation time, and the lobbing errors. These solutions are mainly based on correct...

Authors: Hmaied Shaiek and JeanMarc Boucher

Comparisons of Auditory Impressions and Auditory Imagery Associated with Onomatopoeic Representation for Environmental Sounds

Humans represent sounds to others and receive information about sounds from others using onomatopoeia. Such representation is useful for obtaining and reporting the acoustic features and impressions of actual ...

Authors: Masayuki Takada, Nozomu Fujisawa, Fumino Obata and Shin-ichiro Iwamiya

On the Characterization of Slowly Varying Sinusoids

We give a brief discussion on the amplitude and frequency variation rates of the sinusoid representation of signals. In particular, we derive three inequalities that show that these rates are upper bounded by ...

Authors: Xue Wen and Mark Sandler

Correlation-Based Amplitude Estimation of Coincident Partials in Monaural Musical Signals

This paper presents a method for estimating the amplitude of coincident partials generated by harmonic musical sources (instruments and vocals). It was developed as an alternative to the commonly used interpol...

Authors: JaymeGarciaArnal Barbedo and George Tzanetakis

Efficient Advertisement Discovery for Audio Podcast Content Using Candidate Segmentation

Nowadays, audio podcasting has been widely used by many online sites such as newspapers, web portals, journals, and so forth, to deliver audio content to users through download or subscription. Within 1 to 30 ...

Authors: MN Nguyen, Qi Tian and Ping Xue

Combining Superdirective Beamforming and Frequency-Domain Blind Source Separation for Highly Reverberant Signals

Frequency-domain blind source separation (BSS) performs poorly in high reverberation because the independence assumption collapses at each frequency bins when the number of bins increases. To improve the separ...

Authors: Lin Wang, Heping Ding and Fuliang Yin

Employing Second-Order Circular Suprasegmental Hidden Markov Models to Enhance Speaker Identification Performance in Shouted Talking Environments

Speaker identification performance is almost perfect in neutral talking environments. However, the performance is deteriorated significantly in shouted talking environments. This work is devoted to proposing, ...

Authors: Ismail Shahin

Development of the Database for Environmental Sound Research and Application (DESRA): Design, Functionality, and Retrieval Considerations

Theoretical and applied environmental sounds research is gaining prominence but progress has been hampered by the lack of a comprehensive, high quality, accessible database of environmental sounds. An ongoing ...

Authors: Brian Gygi and Valeriy Shafiro

Adaptive Long-Term Coding of LSF Parameters Trajectories for Large-Delay/Very- to Ultra-Low Bit-Rate Speech Coding

This paper presents a model-based method for coding the LSF parameters of LPC speech coders on a "long-term" basis, that is, beyond the usual 20–30 ms frame duration. The objective is to provide efficient LSF ...

Authors: Laurent Girin

Atypical Speech

Authors: Georg Stemmer, Elmar Nöth and Vijay Parsa

Independent Component Analysis and Time-Frequency Masking for Speech Recognition in Multitalker Conditions

When a number of speakers are simultaneously active, for example in meetings or noisy public places, the sources of interest need to be separated from interfering speakers and from each other in order to be ro...

Authors: Dorothea Kolossa, Ramon Fernandez Astudillo, Eugen Hoffmann and Reinhold Orglmeister

Environmental Sound Perception: Metadescription and Modeling Based on Independent Primary Studies

The aim of the study is to transpose and extend to a set of environmental sounds the notion of sound descriptors usually used for musical sounds. Four separate primary studies dealing with interior car sounds,...

Authors: Nicolas Misdariis, Antoine Minard, Patrick Susini, Guillaume Lemaitre, Stephen McAdams and Etienne Parizet

Determination of Nonprototypical Valence and Arousal in Popular Music: Features and Performances

Mood of Music is among the most relevant and commercially promising, yet challenging attributes for retrieval in large music collections. In this respect this article first provides a short overview on methods...

Authors: Björn Schuller, Johannes Dorfner and Gerhard Rigoll

Exploring the Effect of Differences in the Acoustic Correlates of Adults' and Children's Speech in the Context of Automatic Speech Recognition

This work explores the effect of mismatches between adults' and children's speech due to differences in various acoustic correlates on the automatic speech recognition performance under mismatched conditions. ...

Authors: Shweta Ghai and Rohit Sinha

Animating Virtual Speakers or Singers from Audio: Lip-Synching Facial Animation

Authors: Sascha Fagel, Gérard Bailly and Barry-John Theobald

Semantic Labeling of Nonspeech Audio Clips

Human communication about entities and events is primarily linguistic in nature. While visual representations of information are shown to be highly effective as well, relatively little is known about the commu...

Authors: Xiaojuan Ma, Christiane Fellbaum and Perry Cook

Automatic Recognition of Lyrics in Singing

The paper considers the task of recognizing phonemes and words from a singing input by using a phonetic hidden Markov model recognizer. The system is targeted to both monophonic singing and singing in polyphon...

Authors: Annamaria Mesaros and Tuomas Virtanen

Ageing Voices: The Effect of Changes in Voice Parameters on ASR Performance

With ageing, human voices undergo several changes which are typically characterized by increased hoarseness and changes in articulation patterns. In this study, we have examined the effect on Automatic Speech ...

Authors: Ravichander Vipperla, Steve Renals and Joe Frankel

Wide-Band Audio Coding Based on Frequency-Domain Linear Prediction

We revisit an original concept of speech coding in which the signal is separated into the carrier modulated by the signal envelope. A recently developed technique, called frequency-domain linear prediction (FD...

Authors: Petr Motlicek, Sriram Ganapathy, Hynek Hermansky and Harinath Garudadri

Query-Driven Strategy for On-the-Fly Term Spotting in Spontaneous Speech

Spoken utterance retrieval was largely studied in the last decades, with the purpose of indexing large audio databases or of detecting keywords in continuous speech streams. While the indexing of closed corpor...

Authors: Mickael Rouvier, Georges Linarès and Benjamin Lecouteux

Analysis of the Roles and the Dynamics of Breathy and Whispery Voice Qualities in Dialogue Speech

Breathy and whispery voices are nonmodal phonations produced by an air escape through the glottis and may carry important linguistic or paralinguistic information (intentions, attitudes, and emotions), dependi...

Authors: CarlosToshinori Ishi, Hiroshi Ishiguro and Norihiro Hagita

On the Impact of Children's Emotional Speech on Acoustic and Language Models

The automatic recognition of children's speech is well known to be a challenge, and so is the influence of affect that is believed to downgrade performance of a speech recogniser. In this contribution, we inve...

Authors: Stefan Steidl, Anton Batliner, Dino Seppi and Björn Schuller

Pitch- and Formant-Based Order Adaptation of the Fractional Fourier Transform and Its Application to Speech Recognition

Fractional Fourier transform (FrFT) has been proposed to improve the time-frequency resolution in signal analysis and processing. However, selecting the FrFT transform order for the proper analysis of multicom...

Authors: Hui Yin, Climent Nadeu and Volker Hohmann

Audio Query by Example Using Similarity Measures between Probability Density Functions of Features

This paper proposes a query by example system for generic audio. We estimate the similarity of the example signal and the samples in the queried database by calculating the distance between the probability den...

Authors: Marko Helén and Tuomas Virtanen

Drum Sound Detection in Polyphonic Music with Hidden Markov Models

This paper proposes a method for transcribing drums from polyphonic music using a network of connected hidden Markov models (HMMs). The task is to detect the temporal locations of unpitched percussive sounds (...

Authors: Jouni Paulus and Anssi Klapuri

Automatic Query Generation and Query Relevance Measurement for Unsupervised Language Model Adaptation of Speech Recognition

We are developing a method of Web-based unsupervised language model adaptation for recognition of spoken documents. The proposed method chooses keywords from the preliminary recognition result and retrieves We...

Authors: Akinori Ito, Yasutomo Kajiura, Motoyuki Suzuki and Shozo Makino