John Mourjopoulos

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• Professor Emeritus at University of Patras

Phononic Crystals and Acoustic Metamaterials are an emerging key-enabling technology that is expected to play a significant role in the science and industry of acoustics in the coming years. Until now, the evaluation of acoustic meta-structures has been carried out using traditional methods such as impedance tubes and electromagnetic loudspeakers....

This work proposes a headphone Active Noise Control (ANC) scheme based on the targeted attenuation of sources that are deemed disturbing for listeners. Initially, a listening test determines the disturbance of distinct noises that coexist within background noise based on their class and their direction of arrival. The listening test is based on suc...

A computational framework is proposed for analyzing the temporal evolution of perceptual attributes of sound stimuli. As a paradigm, the perceptual attribute of envelopment, which is manifested in different audio sound reproduction formats, is employed. For this, listener temporal ratings of the envelopment for mono, stereo, and 5.0-channel surroun...

Phononic crystals and acoustic metamaterials are expected to become an important enabling technology for science and industry. Currently, various experimental methods are used for evaluation of acoustic meta-structures, such as impedance tubes and anechoic chambers. Here we present a method for the precise characterization of acoustic meta-structur...

This work studies the spectrum of discrete-time Uniform-sampling pulse width modulation (UPWM) signals originating from stochastic input signals. It demonstrates that for ergodic input sequences of independent and identically distributed random variables, the Discrete Fourier Transform (DFT) of the UPWM signals can be directly estimated from the in...

This work provides an in-depth investigation on the effect of sound diffraction in the acoustics of the ancient theatres, with reference to the theatre of Epidaurus. It is increasingly evident that in such theaters, sound diffraction at the edges of the multiple tiers generates significant source signal amplification. Especially for the distant lis...

Ancient theatres are some of the most significant cultural and architectural masterpieces of the ancient world. The Greeks and Romans developed these spaces for entertainment, political and social commentary, religious rituals and artistic expression. They were built with great attention to acoustics and sightlines. The use of materials such as mar...

This work presents results and models for listener preference to music delivered via different spatial reproduction formats, here via mono, stereo and multichannel (5.1-ch) reproduction. Although this problem has been researched in the past, the current work introduces an elaborate multistage experimental procedure which considers the contribution...

The deposition of femtosecond laser optical energy in gases leads to the emission of secondary electromagnetic and acoustic radiation. These optoacoustic components have a complex nonlinear dependency on the laser beam characteristics, such as the pulse energy, duration, wavelength and the focusing conditions, as well as on the optical and elastic...

Communication Acoustics deals with the fundamentals of those areas of acoustics which are related to modern communication technologies. Due to the advent of digital signal processing and recording in acoustics, these areas have enjoyed an enormous upswing during the last 4 decades. The book chapters represent review articles covering the most relev...

This work presents a novel laser-based optoacoustic transducer capable of reproducing controlled and continuous sound of arbitrary complexity in the air or on solid targets. Light-to-sound transduction is achieved via laser-induced breakdown, leading to the formation of plasma acoustic sources in any desired spatial location. The acoustic signal is...

Reviewing work from diverse scientific fields, this chapter approaches the human aesthetic response to reproduced audio as a process of attraction and efficient (“fluent”) processing for certain auditory stimuli that can be associated with listener pleasure (valence) and attention (arousal), provided that they conform to specific semantic and conte...

Laser-induced breakdown in ambient air by short optical pulses is accompanied by light and sound emission from the excited volume. In this work, these secondary radiative phenomena are jointly studied and their correlation is analyzed and demonstrated both experimentally and analytically. Simultaneous systematic measurements of the light and acoust...

In this presentation, physical and technical aspects of laser-sound generation, a novel technique for reproducing continuous and controlled sound via laser-induced breakdown (LIB) in air, are discussed. Laser-generated sound waves (LIB acoustic pulses) are generally N-shaped and their frequency spectrum extends from the low frequencies of the audib...

Plasma filaments in air induced by femtosecond laser pulses lead to the generation of strong shock waves. This letter presents a systematic study, both experimental and theoretical, of the acoustic radiation by femtosecond laser-generated filaments. A theoretical model is developed based on the experimental results and is used to evaluate the direc...

In this paper a low-complexity method for up-mixing two channel stereo signals for surround sound reproduction is presented. Based on a common stereo signal model, the panning coefficients and azimuth positions of the sources in the stereo mix are estimated. This information is then being used for the stereo signal decomposition and the re-synthesi...

This study employs physical modeling prototypes for investigating acoustic energy harvesting inside closed-box loudspeaker enclosures via a biomimetic mechanism and a piezoelectric transducer. Test results indicate that the proposed device can generate up to 0.34 mW (0.64 μW/cm²) for Lp = 129 dB and 6.58 mW (256 μW/cm²) for Lp = 159 dB inside a lar...

The acoustics of the ancient theatre of Epidaurus have been evaluated in the past via measurements and models. However, the topic still remains open especially with respect to the contributions of the specific architectural elements to the theatre's excellent acoustic performance. Here, the study focuses on 3 novel aspects for a better understandin...

Abstract—The analog voltage-controlled filter used in historical music synthesizers by Moog is modeled using a digital system, which is then compared in terms of audio measurements with the original analog filter. The analog model is mainly borrowed from D’Angelo’s previous work. The digital implementation of the filter incorporates a recently prop...

A method for evaluating different audio systems with respect to their spatial reproduction accuracy is described based on binaural auditory feature classification. The classifier is trained to act as expert listener judging system spatial quality and achieves high accuracy for a reference ideal system under anechoic conditions. The trained classifi...

The principles of acoustical-mechanical coupling between piezoelectric transducers and sound fields are studied for acoustic energy harvesting. The inefficiency of direct coupling is highlighted and an indirect coupling method inspired by the structure and operation of the human middle-ear is introduced. Simulations indicate that the proposed metho...

Byzantine chants constitute a form of monophonic vocal music characterized by lengthy phrases and by musical scales with intervals smaller than the western music semitones. Byzantine churches have extremely long Reverberation Time and their acoustics is dominated by the contributions of the diffuse sound field. Thus, the sound character of Byzantin...

The ancient Greek amphitheaters and roofed odeia represent the earliest examples of acoustics utilized for enhancing theatrical and music performances over large public audiences, often up to 15,000 participants. Such an early achievement, more than 2000 years ago, was possibly crucial for the foundation of these performance-based art forms within...

Byzantine chants constitute a form of monophonic vocal music characterized by lengthy phrases and by musical scales with intervals smaller than the western music semitones. Byzantine churches have extremely long reverberation Time and their acoustics is dominated by the contributions of the diffuse sound field. Thus, the sound character of Byzantin...

A method for the estimation of the direct-to-reverberant-ratio (DRR) from dual-channel microphone recordings without having knowledge of the source signal is proposed. The method is based on previous findings for the statistics of the room transfer function spectral standard deviation and its relationship to the DRR. A novel relationship for the st...

An efficient method for simulating the far field acoustic radiation from resonator panel absorbers is presented following a filter-based modeling approach. This method allows the evaluation of the time and frequency domain response of arbitrary-sized perforated panels in any specific receiver position under free-field and diffuse-field acoustic env...

Byzantine music is typically monophonic and is characterized by (i) prolonged music phrases and (ii) Byzantine scales that often contain intervals smaller than the Western semitone. As happens with most religious music genres, reverberation is a key element of Byzantine music. Byzantine churches/cathedrals are usually characterized by particularly...

A novel method for the estimation of the distance of a sound source from binaural speech signals is proposed. The method relies on several statistical features extracted from such signals and their binaural cues. Firstly, the standard deviation of the difference of the magnitude spectra of the left and right binaural signals is used as a feature fo...

Room reverberation degrades the quality and intelligibility of speech and also reduces the performance of automatic speech-recognition systems. Hence, blind or semi-blind dereverberation methods have been developed, utilising single or multiple input channels. Dereverberation is also important for binaural applications in the context of digital hea...

The performance of recent dereverberation methods for reverberant speech preprocessing prior to Automatic Speech Recognition (ASR) is compared for an extensive range of room and source-receiver configurations. It is shown that room acoustic parameters such as the clarity (C50) and the definition (D50) correlate well with the ASR results. When avail...

The problem of distance estimation by computational methods utilizing binaural information is discussed. Initially, a brief overview is given concerning findings related to the auditory distance perception. Then, several acoustical parameters that depend on the distance between the source and the receiver especially within reverberant rooms are pre...

Extensive results are presented for acoustic and meteorological measurements at the ancient open theatre of Epidaurus. The analysis of the results, illustrates most aspects of the theatre's acoustic properties, indicating the pattern and mechanism for the early reflections, the spectral response of the theatre, aspects of time-frequency response in...

Masks were always used by the actors of the ancient Greek theatrical plays. Their theatrical function indicates an integral connection between the performers, the plot and the acoustics of the open-air ancient theatres which has not been explored in the past. Although the exact form, shape and material of such masks can be only deduced from limited...

In this study, the results of the ongoing work of the authors on the topic of the analysis of the acoustical environment from reverberant signals will be presented. Initially, some theoretical aspects on the relationships of the statistical quantities of the room transfer functions and the reverberant signals will be given. Then, the way that these...

In this paper an implementation for a small-sized subwoofer system is proposed, utilizing three 4-inch drivers. The low frequency reproduction approach adopted here is based on narrow bandwidth loudspeaker design with high quality factors. The desired resonance frequency of each loudspeaker is obtained by increasing the cone mass, in order to minim...

In this paper, the implementation of a transient-aware, discrete-time audio processor is presented. The processing takes place in the Short Time Fourier Transform (STFT) domain, using a variant of the well known overlap-add technique. The proposed method identifies the presence of transients in the input signal and constantly adapts the analysis-sy...

Nowadays, numerous daily life applications related to wireless networks, sensors, low power electronics, biomedical applications etc. demand small amount of electrical power for their operation. These applications are rapidly increasing and bringing to the fore the need for small scale electrical power generation. To this end, strong efforts are be...

The close-microphone technique is widely used in modern sound engineering practice. It is mainly used to minimize the effect of leakage and room acoustics on the received signal. In this work, the properties of the close-microphone response are investigated from a signal processing point of view, through the respective frequency domain statistical...

Microphone leakage is one of the most prevalent problems in audio applications involving multiple instruments and multiple microphones. Currently, sound engineers have limited solutions available to them. In this paper, the applicability of two widely used signal enhancement methods to this problem is discussed, namely blind source separation and n...

Despite recent advances in wireless networking technology, real-time streaming of CD-quality digital audio remains a challenging topic. In this work, a set of applications following the server-client model was developed, facilitating the transmission and playback of PCM-coded audio over wireless links. The implementation is based on typical Persona...

Real time, multichannel audio content delivery over the air is expected to significantly simplify the interconnection complexity required for setting up typical home theater applications. However, despite the technological advantages of wireless networking standards related to high transmission rates and Quality-of-Service support, a number of issu...

Thermoacoustic transduction is the transformation of thermal energy fluctuations into sound. Devices fabricated by appropriate materials utilise such a mechanism in order to achieve acoustic wave generation by direct application of an electrical audio signal and without the use of any moving components. A thermoacoustic transducer causes local vibr...

Loudspeaker Arrays driven by digital bitstreams are direct digital-signal to acoustic transducers, usually comprising of a digital signal processing module driving actuators. Current research efforts are focusing on topologies directly driven by multi-bit digital bitstreams. In this work, the above investigations are extended to the case of using 1...

Digital loudspeaker arrays currently are based on small moving-coil speakers to reconstruct acoustic signals out of binary audio streams. An overview of significant performance issues for such systems is given here to explain frequency response and speaker discrete transition rates due to the digital data. Detailed simulations provided comparisons...

Digital audio filters with time-varying parametric features are analyzed, and their realization via recursive difference equations with updated coefficients is examined. The errors and distortions occurring during parameter transitons are evaluated, and their audible components are estimated using a computational model of the auditory masking mecha...

Significant measurable audible distortions are generated during real-time room acoustics dereverberation, which are far greater in magnitude than those expected from identical simulated (off-line) dereverberation experiments. Long inverse filters worsen this effect. Possible mechanisms for such discrepancy between simulated and real-time tests are...

A novel technique for the equalization of loudspeakers and other audio system components is presented. The method is based upon digital prefiltering of the audio signals to compensate for distortions, with the filter designed in such a way as to give spectral, phase, and transient equalization. Results are presented for computer simulations and for...

The well-known property of the spectral standard deviation of Room Transfer Functions (RTFs), that is, its convergence to 5.57 dB, is extended to reverberant Binaural Room Transfer Functions (BRTFs). The BRTFs are related to the anechoic Head Related Transfer Functions (HRTFs) and the corresponding RTFs. Consequently, the statistical properties of...

Theatre masks were fundamental elements of the ancient Greek theatre tradition, having a dramatic impact on the artistic performance. Apart from the obvious change of the visual appearance of the actors, the masks also altered the acoustic characteristics of their voices. Therefore, both from the spectator's and the actor's point of view these mask...

A method to detect the distance of a speaker from a single microphone in a room environment is proposed. Several features, related to statistical parameters of speech source excitation signals, are introduced and are shown to depend on the distance between source and receiver. Those features are used to train a pattern recognizer for distance detec...

A semi-blind framework for the suppression of late speech reverberation is presented. The method is based on spectral subtraction and utilizes a simple recorded handclap to estimate the Power Spectral Density (PSD) of late reverberation. A statistical analysis of measured Room Impulse Responses (RIRs) and recorded handclaps demonstrates the suffici...

A spectral subtraction method for dereverberation at multiple speaker positions utilizing a single Room Impulse Response (RIR) measurement is presented. A running kurtosis approach is applied in order to define the RIR early/late reflections boundary. Then, the power spectrum of the late reverberant part of the measured RIR along with the excitatio...

The spectral magnitude of Room Transfer Functions (RTFs) and signals (anechoic and reverberant) is analysed using statistical quantities, such as the standard deviation and the kurtosis. These measures are examined for RTFs recorded at dierent distances between source and receiver and their dependency on distance is modeled using polynomial express...

Adapting single channel dereveverberation techniques for binaural processing is not trivial. Apart from the challenging task of reducing reverberation without introducing audible artifacts, binaural dereverberation methods should also at least preserve the Interaural Time Dierence (ITD) and Interaural Level Dierence (ILD) cues as it has been shown...

In previous work of the authors, the spectral magnitude of room transfer functions (RTFs) was analyzed using histograms and statistical quantities (moments), such as the kurtosis and skewness. In this work, the above analysis is extended to binaural room impulse responses (BRIRs) and the dependence of the statistical measures on the room acoustical...

An algorithm for joint suppression of noise and reverberation from speech signals is presented. The method requires a handclap recording that precedes speech activity. A running kurtosis technique is applied in order to extract an estimation of the late reflections of the room impulse response from the clap while a moving average filter is employed...

Single-channel spectral subtraction algorithms are commonly used to suppress late reverberation. A binaural extension of such methods, apart from suppressing reverberation without introducing processing artifacts, should also preserve the signal's binaural localization cues. Here, three state-of-the-art spectral subtraction dereverberation algorith...

In this work a simple and effective method is proposed to detect time intervals where only a single source is active (solo intervals) for multiple microphone, multiple source settings commonly encountered in audio applications, such as live sound reinforcement. The proposed method is based on the short term energy ratios between all available micro...

Although dereverberation can be useful in many audio applications, such techniques often introduce artifacts that are unacceptable in audio engineering scenarios. Recently, the authors have proposed a novel dereverberation approach, suitable for both speech and music signals, based on perceptual reverberation modeling. Here, the method is fine-tune...

A perceptually motivated signal-dependent metric for modeling perceived alterations in audio signals caused by reverberation, the reverberation masking index (RMI), is proposed. Unlike existing descriptions and measures the proposed RMI is varying dynamically with each signal's evolution in a way that appears to be compliant to the perceived altera...

A blind method for suppressing late reverberation from speech and audio signals is presented. The proposed technique operates both on the spectral and on the sub-band domains employing a single input channel. At first, a preliminary rough clean signal estimation is required and for this, any standard technique may be applied; however here the estim...

A method to estimate the distance of a speaker from a single microphone in a room environment is studied. Several features, related tostatistical parameters of speech source excitation signals, areintroduced and are shown to depend on the distance between sourceand receiver. Those features are used to train a pattern recognizerfor distance estimati...

The problem of acoustic source separation is addressed in an audio application context, especially when considering close-microphone applications and the effects of room acoustics. The framework of blind source separation (BSS) is used here to formally describe the problem, and solutions derived from this framework are examined for the case of sepa...

This work examines statistical quantities (i.e. kurtosis, skewness) of room transfer functions and audio signals (anechoic, reverberant, speech, music). Measurements are taken under various reverberation conditions in different real enclosures ranging from small office to a large auditorium and for varying source -receiver positions. Here, the stat...

The identification of room impulse responses is very important in many audio signal processing applications. Typical system identification methods require access to the original source signal which is seldom available in practice, while blind system identification methods require prior knowledge of the statistical properties of the source signal wh...

The suppression of late reverberation by spectral subtraction tends to degrade disproportionally low-level signal regions and signal transients. This work proposes two novel relaxation criteria that can constrain such problems in a signal-dependent fashion. These criteria were found to improve the performance of state-of-the-art late reverberation...

The analysis of the sound field generated by a Helmholtz resonators panel is described by a Finite Elements Method. The resonance frequency, absorption and Q factor of the panel was evaluated at different positions for a sound source and different geometries of the panel mainly over the frequency range from 20Hz to 100Hz. The results showed that as...

Small loudspeakers with low force factors (Bl) behave as narrow bandwidth filters with high quality factors and can be tuned at low resonance frequencies. It is possible to optimize low frequency reproduction (in the sub-woofer range) using a number of such drivers. This study employs a set of such low-Bl loudspeakers with different eigenfrequencie...

An improved spectral subtraction algorithm for suppressing late reverberation is presented. Existing spectral subtraction methods, process similarly all frames of the reverberant speech signal. This work introduces two criteria for the identification of the speech frames that do not contain significant late reverberation power. The subtraction is a...

The proposed model derives time-frequency maps to estimate perceived alterations due to reverberation in stereo audio signals reproduced in rooms. These alterations relate to monaural masking due to reverberant decay, derived via a computational auditory masking model and to inter-channel cues for the formation of the spatial position of the aural...

In this paper we study the problem of estimating the distance of a sound source from a single microphone recording in a room environment. The room effect cannot be separated from the problem without making assumptions about the properties of the source signal. Therefore, it is necessary to develop methods of distance estimation separately for diffe...

Ancient Greek / Roman odeia were semi-enclosed theaters that currently survive without their original roof sections. The work compares the acoustics of the well-preserved Herodes Odeion in its current open-air form to a detailed acoustic model reconstruction of its original roofed version and illustrates the significant differences in acoustics bet...

A novel technique of perceptually-motivated signal dependent audio mixing is presented. The proposed Hierarchical Perceptual Mixing (HPM) method is implemented in the spectro-temporal domain; its principle is to combine only the perceptually relevant components of the audio signals, derived after the calculation of the minimum masking threshold whi...

Digital loudspeaker arrays (DLAs) are electroacoustic transducers reconstructing acoustic signals out of binary audio streams, usually consisting of a digital signal processing module driving multiple actuators (small loudspeakers). In this work a DLA prototype was implemented using an all-digital audio chain driven by 1-bit sigma-delta signals. Th...

A new method for blind estimation and suppression of late reverberation of speech signals is presented. The proposed algorithm consists of two steps. In a first step, the reverberation time is blindly determined from the reverberant signal. Then, an approximation of the power spectrum of late reverberation is subtracted from the power spectrum of t...

Many current research efforts focus on alternative electroacoustic transduction devices having no moving parts, in order to achieve sufficient audio performance from compact solid state devices. Thermoacoustic loudspeakers are transducers based on the conversion of AC current signals to thermal energy, causing a local fluctuation of air pressure wh...

For some time now, statistical analysis has been a valuable tool in analyzing room transfer functions (RTFs). This work examines existing statistical time-frequency models and techniques for RTF analysis (e.g., Schroeder's stochastic model and the standard deviation over frequency bands for the RTF magnitude and phase). RTF fractional octave smooth...

Audio delivery and reproduction for home or professional applications may greatly benefit from the adoption of digital wireless local area network (WLAN) technologies. The most challenging aspect of such integration relates the synchronized and robust real-time streaming of multiple audio channels to multipoint receivers, for example, wireless acti...

Although uniformly sampled pulse width modulation (UPWM) represents a very efficient digital audio coding scheme for digital-to-analog conversion and full-digital amplification, it suffers from strong harmonic distortions, as opposed to benign non-harmonic artifacts present in analog PWM (naturally sampled PWM, NPWM). Complete elimination of these...

Recent developments in Digital Loudspeakers have resulted in the introduction of Digital Transducer Arrays (DTA). In most implementations, DTA loudspeakers are driven by PCM encoded audio signals, usually resampled and requantised to an appropriate number of bits, in accordance to the number of the transducers constituting the DTA topology. However...

From reverberated audio signals and using as reference the input (anechoic) audio, a number of distortion maps are extracted indicating how room reverberation distorts in time-frequency scales, perceived features in the received signal. These maps are simplified to describe the monaural time-frequency / level distortions and the distortion of the s...

The recent growth of home networking applications has raised the requirement for cable-free audio streaming. In this work, the "BlueDAT" digital audio platform is presented, which employs the Bluetooth wireless technology for real-time, CD-like quality, stereo reproduction. It is shown that, despite the strict throughput limitations imposed by the...

Real-time digital wireless playback of CD-quality audio in multipoint setups using quality of service enhancements is analyzed and evaluated in this work. A novel methodology is introduced for simulating wireless digital audio delivery as well as for theoretically deriving the playback distortions. This methodology allows the accurate wireless digi...

A novel signal-dependent approach is followed here for modeling perceived distortions due to reverberation in audio signals. The method describes perceived monaural time-frequency and level distortions due to reverberation which depend on the reproduced signal's evolution. A Computational Auditory Masking Model (CAMM) is employed, using as inputs t...

Improvement of sound reproduction in rooms by signal processing techniques has been studied seriously for more than twenty years. Such methods are called for example dereverberation, room (response) equalization, and room (response) correction. Understanding the limitations and quality criteria for such methods is gradually achieved through studies...