Bruno Masiero

The electrical engineering bachelor course at the University of Campinas, Brazil, offers 13 areas of specialization, one of them being Sound Engineering. This specialization provides students with basic knowledge on engineering tools and applications related to sound and audio. To be eligible for this certificate the student has to participate in a total of 16 credit hours of courses related to classical acoustics, digital signal processing for audio, and musical theory (musical notation and perception). One of the core courses in this program is "Analysis and Synthesis of Musical Signals". At the end of this course, the student should be able to design a system for audio signal analysis and synthesis, and properly adjust the system's parameters for a particular purpose. In the first semester of 2016, this course was reorganized based on "active learning" and "project-based learning" methodologies. In this manuscript, we discuss the motivations for ...

This article describes the design, implementation, and experiences with AcMus, an open and integrated software platform for room acoustics research, which comprises tools for measurement, analysis, and simulation of rooms for music listening and production. Through use of affordable hardware, such as laptops, consumer audio interfaces and microphones, the software allows evaluation of relevant acoustical parameters with stable and consistent results, thus providing valuable information in the diagnosis of acoustical problems, as well as the possibility of simulating modifications in the room through analytical models. The system is open-source and based on a flexible and extensible Java plug-in framework, allowing for cross-platform portability, accessibility and experimentation, thus fostering collaboration of users, developers and researchers in the field of room acoustics.

The sound absorption of materials is traditionally measured in laboratory condition with one of two methods: random incidence in a reverberant chamber (ISO 354) or normal incidence in an impedance tube (ISO 10534). Nevertheless, there are some materials that cannot be measured in the lab, e.g., road surfaces, which are recommended to be measured in-situ with the use of a single microphone (ISO 13472). The latter method is based on time-windowing the measured impulse response to compare the incident and reflected wave components. Depending on the measurement setup, the size of the window may result in degraded measurement quality, specially at low frequencies. With the intention to alleviate this effect we evaluate three blind deconvolution approaches. These methods blindly estimates the response of the sound source, resulting in a cleaner (possibly more sparse) measurement that can than be used to estimate sound absorption. We compare the results of the aforementioned methods with f...

Beamforming techniques are commonly applied to signals captured by sensor arrays to enhance signals received from desired directions while reducing background noise and localized interference. Where the directions of the desired and interfering sources are known, this knowledge, combined with assumptions on the background noise characteristics, is used to derive the beamformer coefficients for each sensor. Usually, this is done by optimizing the second-order statistics of the beamformer response, e.g., minimizing the energy of the output signal while preserving the signals from desired directions. The beamformer coefficients are independently derived for each discrete frequency, as an approximation to the true broadband response. Hereby, the complex inter-frequency interactions, e.g., due to windowing and spectral aliasing, are not modelled, leading to sub-optimal filter characteristics. Furthermore, for standard designs, the mainlobe also narrows with frequency, leading to a non-un...

Headphones must always be adequately equalized when used for reproducing binaural signals if they are to deliver high perceptual plausibility. However, the transfer function between headphones and ear drums (HpTF) varies quite heavily with the headphone fitting for high frequencies, thus even small displacements of the headphone after equalization will lead to irregularities in the resulting frequency response. Keeping in mind that irregularities in the form of peaks are more disturbing than equivalent valleys, a new method for designing headphone equalization filters is proposed where not the average but an upper variance limit of many measured HpTFs is inverted. Such a filter yields perceptually robust equalization since the equalized frequency response will, with high chance, differ from the ideal response only by the presence of valleys in the high frequency range.

The problem of acoustic scene description with sensor ar- rays is to determine the number and location of (usually few) sound sources present in a (possibly noisy) sound scene from measurements of the wave field with a mi- crophone array. Conventional beamforming is the most usual method to extract the sources’ direction-of-arrival and emitted signal, even though it is characterized by low spatial resolution. The compressive beamforming (CB) method asserts that spatially sparse signals can be recovered from arrays with reduced number of sensors by solving a convex minimizati- on problem. However, despite the fact that the compressi- ve sensing framework applied in CB offers computational efficiency compared to other sparsity promoting methods, its iterative algorithm is still very time consuming when compared with conventional beamforming. In the quest for a real-time implementation of CB, we present the Kronecker Array Transform (KAT) to speed up the bott- leneck of the CB algorith...

This paper describes the results obtained in an under-grad project in the area of acoustical measuring. In this project, a research about the various acoustic impulse response measurement systems for small room was made. A research about room acoustical parameters, as well as about impulse response processing methods for its derivation, was also done. As a result of this project, a acoustic impulse response measurement system was developed. Resumo. Neste projeto foi feita uma revisão das diferentes técnicas de medição da resposta impulsiva acústica para salas de pequeno porte. Foi feita também uma revisão dos parâmetros acústicos para salas, assim como métodos de processamento da resposta impulsiva para sua obtenção dos mesmos. Como resultado final do projeto, foi desenvolvido um sistema de medição da resposta impulsiva acústica. 1. Métodos de Obtenção da Resposta Impulsiva A resposta impulsiva acústica é uma função temporal da pressão sonora de um espaço acústico, que resulta da ex...

One of the main challenges to a successful ANC application on PW is determining the quantity and location of control sources (CS) and error sensors (ES). The definition of these parameters is not an obvious task, and there are yet no closed-form expressions that yield their optimal value, which depends on the sound field generated by the PW, the frequency band where attenuation is desired and many other factors.

This is a lab report paper about the state of affairs in the computer music research group at the School of Electrical and Computer Engineering of the University of Campinas (FEEC/Unicamp). This report discusses the people involved in the group, the efforts in teaching and the current research work performed. Last, it provides some discussions on the lessons learned from the past few years and some pointers for future work.

In recent years, many techniques for sparse signal recovery have aroused interest due to their applicability in the area of acoustic source localization. Among them is the Sparse Learning via Iterative Minimization (SLIM) method. However, most of these techniques assume that the signals emitted by the sources have narrow band, which in practice does not fit usual acoustical applications. The wideband extension of SLIM (WB-SLIM) is able to handle broadband sources and promotes the spatial sparseness of the results. This paper aims to evaluate the performance of WB-SLIM in different acoustic imaging scenarios. We analyzed the performance of WB-SLIM in a 2D scanning region, both parallel and perpendicular to the microphone array. The results illustrate how the quality of the acoustic images in both proposed scenarios, evaluated as the source localization error rate and through the comparison between recovered and original time signals, decreases as the number of sources composing the sound field increases and the signal-to-noise ratio decreases.

ABSTRACT Measuring spatial features of sound sources and receivers is typically a time consuming task, especially when a high spatial resolution is required, as independent measurements have to be conducted for each measured direction. A speed-up in measurement time can be achieved with parallel measurement techniques using arrays of sound sensors or sources. For linear and time-invariant systems only loose restrictions are claimed for the excitation signal and the measurement method. Nevertheless, when measuring a sound receiver, e.g., directional microphones, the signals emitted by the multiple sound sources must be separable. Acoustic systems can be treated as linear systems for low input levels. However, when it comes to moderate levels, loudspeakers show non-linear behavior that cannot be neglected. To conduct a parallelized measurement technique at these levels the multiple exponential sweep method has recently been introduced to measure the acoustic transfer characteristics with weakly non- linear sound sources by using exponential sweeps. This method decreases the measurement time compared to sequential measurements. However, compared to the ideal linear case, the measurement duration is increased due to occurring harmonic impulse responses. A novel generalized overlapping strategy for these sweeps is proposed considering the length of each harmonic impulse response and, additionally, the temporal structure of the desired impulse responses measured in anechoic environments. It is shown that the resulting optimized multiple exponential sweep method can yield even shorter measurement times than the original method.

When comparing the binaural presentation of sound events, clear differences regarding overall quality between headphones and crosstalk-canceled loudspeaker reproduction are reported. In particular, “in-head localization” occurs more often with headphone than with loudspeaker reproduction. It seems that listeners are less sensitive to loudspeaker distortion, either sound-wise or spatially, then to headphone distortion. The reason for this phenomena is a not sufficiently precise equalization of the head-headphone system. When stimuli are played through headphones, an alteration of the sound signal in the ear occurs caused by resonances and nonlinearities in the headphones and also by the coupling of the headphones to the ear canal. To provide a defined interface for psychoacoustic listening tests with headphones, the headphones must be adequately equalized.

ABSTRACT The aim of this study was to examine auditory selective attention in language switching. To this end, we used a novel variant of dichotic selective listening and examined language comprehension. In our task, subjects had to respond selectively to one of two simultaneously presented auditory stimuli (number words in German and English), always spoken by a female speaker, by performing a numerical size categorization (smaller vs. larger than five). The language the subjects had to respond to could switch from trial to trial either unpredictably (i.e., cued; Experiment 1) or predictably (Experiment 2). We found clear performance costs with language switches as compared to language repetitions. Moreover, incongruent numerical categories in competing auditory stimuli produced interference and increased error rates, suggesting continued processing of task-irrelevant information.

This paper describes the results obtained in an under-grad project in the area of acoustical measuring. In this project, a research about the various acoustic impulse response measurement systems for small room was made. A research about room acoustical parameters, as well as about impulse response processing methods for its derivation, was also done. As a result of this project, a acoustic impulse response measurement system was developed. Resumo. Neste projeto foi feita uma revisão das diferentes técnicas de medição da resposta impulsiva acústica para salas de pequeno porte. Foi feita também uma revisão dos parâmetros acústicos para salas, assim como métodos de processamento da resposta impulsiva para sua obtenção dos mesmos. Como resultado final do projeto, foi desenvolvido um sistema de medição da resposta impulsiva acústica.

The sound-source localization provided by a crosstalk cancellation (CTC) system depends on the head-related transfer functions (HRTFs) used for the CTC filter calculation. In this study, the horizontal- and sagittal-plane localization performance was investigated in humans listening to individualized matched, individualized but mismatched, and non-individualized CTC systems. The systems were simulated via headphones in a binaural virtual environment with two virtual loudspeakers spatialized in front of the listener. The individualized mismatched system was based on two different sets of listener-individual HRTFs. Both sets provided similar binaural localization performance in terms of quadrant, polar, and lateral errors. The individualized matched systems provided performance similar to that from the binaural listening. For the individualized mismatched systems, the performance deteriorated, and for the non-individualized mismatched systems (based on HRTFs from other listeners), the...

Binaural stimuli presented via headphones need to be plausible in localization and sound coloration for a successful reproduction of an acoustic scene, especially for experiments on auditory selective attention. The goal is to provide artificially generated acoustic scenes in a way that the difference between a real situation and an artificially generated situation has no influence in psychoacoustic experiments. The quality and reliability of binaural reproduction via headphones comparing two different microphone setups (miniature microphone in open dome and ear plug) used for individualized head-related transfer functions and headphone transfer function measurements is analyzed. Listening tests are carried out focusing on authenticity, naturalness, and distinguishability in a direct comparison of real sources and binaural reproduction via headphones. Results for three different stimuli (speech, music, pink noise) are discussed. Furthermore, approaches to perform experiments on audi...

ABSTRACT Dynamic crosstalk cancellation (CTC) systems commonly find use in immersive virtual reality (VR) applications. Such dynamic setups require extremely high filter update rates, so filter calculation is usually performed in the frequency-domain for higher efficiency. This paper proposes a general framework for the calculation of dynamic CTC filters to be used in immersive VR applications. Within this framework, we introduce a causality constraint to the frequency-domain calculation to avoid undesirable wrap-around effects and echo artifacts. Furthermore, when regularization is applied to the CTC filter calculation, in order to limit the output levels at the loudspeakers, noncausal artifacts appear at the CTC filters and the resulting ear signals. We propose a global minimum-phase regularization to convert these anti-causal ringing artifacts into causal artifacts. Finally, an aspect that is especially critical for dynamic CTC systems is the filter switch between active loudspeakers distributed in a surround audio-visual display system with 360 $^circ$ of freedom of operator orientation. Within this framework we apply a weighted filter calculation to control the filter switch, which allows the loudspeakers’ contribution to be windowed in space, resulting in a smooth filter transition.

ABSTRACT The aim of this study was to examine auditory selective attention in language switching. To this end, we used a novel variant of dichotic selective listening and examined language comprehension. In our task, subjects had to respond selectively to one of two simultaneously presented auditory stimuli (number words in German and English), always spoken by a female speaker, by performing a numerical size categorization (smaller vs. larger than five). The language the subjects had to respond to could switch from trial to trial either unpredictably (i.e., cued; Experiment 1) or predictably (Experiment 2). We found clear performance costs with language switches as compared to language repetitions. Moreover, incongruent numerical categories in competing auditory stimuli produced interference and increased error rates, suggesting continued processing of task-irrelevant information.