Optimism scale: Evidence of psychometric validity in two countries and correlations with personality

Optimism can be defined as the hope that something good is going to happen in the future. It is a relevant construct in the study of happiness, and is associated with a range of variables, including subjective well-being, reduced risk of suicidal ideation, quality of social relationships, and a healthier lifestyle. However, current measures of optimism were criticized regarding their structure and reliability. To address these limitations, Pedrosa et al. (2015) proposed a new scale of dispositional optimism that was originally published in Spanish. In the present research, we aimed to provide further psychometric evidence of the 9-item Optimism Scale in the United Kingdom (N = 325) and Brazil (N = 421). Exploratory and confirmatory factor analyses in both countries were consistent with the original findings, supporting the unifactorial structure. Item Response Theory revealed good discrimination, level of difficulty, and informativeness of the items. Further, we found good reliability estimates of the scale, full factorial invariance across participants' gender and partial invariance across countries, and positive correlations with all Big-5 personality traits. In sum, our findings suggest that the dispositional Optimism Scale is a psychometrically adequate measure that can be used cross-culturally

Microphone and loudspeaker arrays for room acoustic measurements

A multichannel spherical speaker array allows, together with a spherical microphones array, the measurement of the MIMO (Multiple Input Multiple Output) acoustic impulse response of an environment capturing meaningful information about propagation of sound between source an receiver. The mathematical framework for extracting arbitrary directivity virtual microphones from real microphones array signals is recalled and the application of the same method to the speakers array to generate arbitrary directivity source is presented. A convenient solutions for the construction and calibration of speakers spherical array for measurement purposes is illustrated. The postprocessing technique developed to compute and visualize acoustic path between source and receiver from measured MIMO impulse response is discussed. Real word results from measurement in a small theater are shown

A novel 32-speakers spherical source

The construction and test of a novel compact spherical source equipped with 32 individually driven 2" loudspeakers is presented. The new sound source is designed for making room acoustics measurements, emulating the directivity pattern of various music instruments or human talkers and singers. The 32 signals feeding the loudspeakers can be obtained by three different approaches: a set of High Order Ambisonics coefficients computed for emulating the polar pattern of a fixed directivity source a set of SPS (Spatial PCM Sampling) signals recorded around a real source, employing a corresponding set of 32 microphones placed on a sphere surrounding the real source, a matrix of FIR filters, designed employing a mathematical theory almost identical to the one developed for creating virtual microphones from a spherical microphone array [1] The presentation will show details of the construction of the new loudspeaker array, and the results of the first tests performed for evaluating the capability of creating arbitrary polar radiation patterns

Measuring Spatial MIMO Impulse Responses in Rooms Employing Spherical Transducer Arrays

The paper presents a new measurement method aimed to characterise completely the sound propagation form a point to another point inside a room, taking into account the directionality of the source and of the receiver. The method makes use of two spherical arrays of transducers, almost uniformly scattered on the surface of a rigid sphere, which can synthesize arbitrary polar patterns. The paper describes the beamforming method employed for synthesizing the required polar patterns over a wide frequency range, and how to process the results in various ways: graphical mapping of sound reflections inside the room, reconstructing the trajectories of such reflections and auralization, when for the first time both source and receiver can be allowed to freely rotate during the test

SPATIAL SOUND RECORDING WITH DENSE MICROPHONE ARRAYS

Multichannel recordings are usually performed by means of microphone arrays. In many cases "sparse" and discrete microphone arrays are used, where each microphone is employed for capturing one of the channels, which in turn is routed to one loudspeaker. However, also the usage of "dense" microphone arrays has a long history, dating back to the first MS-matrixed microphones setups and passing through the whole Ambisonics saga. A dense microphone array is employed differently from a sparse array: each channel is obtained by a combination of the signals coming from all the capsules, by means of different matrixing and filtering approaches. And similarly, each loudspeaker feed results from a re-matrixing of all the transmitted channels. This paper is the third of a series: in the previous two [1,2] a numerical method for computing a matrix of FIR filter was employed for processing the microphone signals (encoding, [1]) and for computing the speaker feeds (decoding, [2]). In this third paper, the same numerical approach is extended to intermediate processing (rotation, zooming, stretching, spatial equalization, etc.): hence we have now a general meta-theory, providing a unique framework capable of processing the signals for any kind of dense microphone array, providing any kind of intermediate manipulation, and finally projecting the signal to every kind of loudspeaker arrays. The same framework can operate according to different standards and formats, including A-format (raw signals), B-format (High Order Ambisonics signals), G-format (speaker feeds) and P-format (Spatial PCM Sampling signals), and can be used for converting freely among them. Experimental results are presented, including "traditional" tetrahedral probes, a commercial spherical microphone array, and two newly-developed massive microphone arrays developed by the authors, a cylindrical and a planar array, both incorporating 32 high-quality condenser microphones and a panoramic video camera

A Spherical Microphone Array For Synthesizing Virtual Directive Microphones In Live Broadcasting And In Post Production

The paper describes the theory and the first operational results of a new multichannel recording system based on a 32- capsules spherical microphone array. Up to 7 virtual microphones can be synthesized in real-time, choosing dynamically the directivity pattern (from standard cardioid to 6th-order ultradirective) and the aiming. A graphical user’s interface allows for moving the virtual microphones over a 360-degrees video image. The system employs a novel mathematical theory for computing the matrix of massive FIR filters, which are convolved in real time and with small latency thanks to a partitioned convolution processor

Spatial Pcm Sampling: A New Method For Sound Recording And Playback

This paper presents the mathematical and physical framework of a new technology, named SPS (Spatial PCM Sampling): it is the equivalent, in a two-dimensional spherical-coordinate space, of the traditional PCM representation of a waveform (in the one-dimensional time domain). It is nowadays possible to record an SPS multichannel stream (also called P-format) by processing the signals coming from massive microphone arrays, now widely employed in the broadcasting industry and in research labs. Some types of sound processing are easy when operating on P-format signals; some, indeed, require more work. At playback, it is possible to drive loudspeaker arrays of arbitrary shape and complexity, providing in general better spatial accuracy than competing well known methods, such as Ambisonics or WFS

Monitoring water stress in Mediterranean semi-natural vegetation with satellite and meteorological data

In arid and semi-arid environments, the characterization of the inter-annual variations of the light use efficiency ε due to water stress still relies mostly on meteorological data. Thus the GPP estimation based on procedures exclusively driven by remote sensing data has not found yet a widespread use. In this work, the potential to characterize the water stress in semi-natural vegetation of three spectral indices (NDWI, SIWSI and NDI7) – from MODIS broad spectral bands – has been analyzed in comparison to a meteorological factor (Cws). The study comprises 70 sites (belonging to 7 different ecosystems) uniformly distributed over Tuscany, and three eddy covariance tower sites. An operational methodology, which combines meteorological and MODIS data, to characterize the inter-annual variations of ε due to summer water stress is proposed. Its main advantage is that it relies on existing series of meteorological data characterizing each site and allows calculating a typical Cws profile that can be “updated” (C∗ ws) for the actual conditions using MODIS spectral indices. The results confirm that the modified C∗ ws can be used as a proxy of water stress that does not require concurrent information on meteorological data.JRC.H.3-Forest Resources and Climat

A Modular, Low Latency, A2B-based Architecture for Distributed Multichannel Full-Digital Audio Systems

Despite the increasing demand for multichannel audio systems, existing solutions are still mainly analog or audio-over-IP based, leading to well-known limitations: bulky wiring, high latency (0.5–2 ms), and expensive devices for protocol stack management. This paper presents a cost-effective, low latency, full-digital solution that overcomes all the previously mentioned problems. The proposed architecture is based on the new Automotive Audio Bus (A2B) protocol. It guarantees deterministic latency of 2 samples, 32 downstream/upstream channels over a single Unshielded Twisted Pair (UTP) cable and phase-aligned signals. A single A2B chip is required for each node, reducing dramatically the system cost. The developed architecture is composed by a main board and an A2B network. The main board handles up to 64 channels, and it converts standard protocols usually employed for audio signal delivery, such as AES10, AVB and AES67, into A2B streams and vice versa. The A2B network can include a series of devices, for instance power amplifiers, codecs, DSPs, and transducers. There are many application examples including, but not limited to, transducer arrays (e.g., microphone, loudspeaker, accelerometer arrays), audio distribution in meeting rooms, Wave Field Synthesis (WFS), Ambisonics immersive audio systems and Active Noise Control (ANC). A modular and portable WFS system was developed employing the above-described architecture. It is based on eight channels soundbars, which can be daisy-chained in reconfigurable geometries and featuring up to 192 channels