Acoustic characterisation of cinema screens

As a consequence of cinema screens being placed in front of screen-speakers, a reduction in sound quality has been noticed. Cinema screens not only let the sound go through them, but also absorb a small amount of it and reflect the sound which impacts on the screen to the back, coming forward again in case it impacts on the loudspeaker. This backwards reflection in addition to the signal coming from the loudspeaker can lead to constructive or destructive interference at certain frequencies which usually results in comb filtering. In this project, this effect has been studied through researching amongst various data sheet provided by different manufacturers, acoustical measurements completed in the large anechoic chamber of the ISVR and some theoretical models developed with MatLab software. If results obtained with MatLab are accurate enough in comparison to the real measurements taken in the anechoic chamber this would lead to a good way to predict which would be the attenuation added to the system at each frequency, given that not all manufacturers provide an attenuation curve, but only an average attenuation. This average attenuation might be useless as sound waves have different wavelengths and its propagation through partitions varies. In fact, sound is composed by high and low frequencies, where high frequencies are characterised by a small wavelength which is usually easier to attenuate than low frequencies that characterised by bigger wavelengths. Furthermore, this information would be of great value to both screen manufacturers, who could offer a much more precise data in their data sheets; and customers, who would have a great amount of information to their disposal before purchasing and installing anything in their cinemas, being able to know by themselves which screen or loudspeaker should be best to meet their expectative. RESUMEN. La aparición de la digitalización de las bandas sonoras para las películas hace posible la mejora en la calidad de sonido de los cines. Sin embargo, un aspecto a tener en cuenta en esta calidad del sonido es la transmisión de éste a través de la pantalla, ya que normalmente tras ella se encuentran situados los altavoces. Las propiedades acústicas varían dependiendo del tipo de pantalla que se utilice, además de haber poca información a la que acceder para poder valorar su comportamiento. A lo largo de este proyecto, se analizan tres muestras de pantallas distintas donadas por distintos fabricantes para poder llegar a la conclusión de dependiendo del tipo de pantalla cuál es la distancia óptima a la que localizar la pantalla respecto al altavoz y con qué inclinación. Dicho análisis se realizó en la cámara anecoica del ISVR (University of Southampton) mediante la construcción de un marco de madera de 2x2 m en el que tensar las pantallas de cine, y un altavoz cuyo comportamiento sea el más similar al de los altavoces de pantalla reales. Los datos se captaron mediante cuatro micrófonos colocados en posiciones distintas y conectados al software Pulse de Brüel & Kjær, a través del cual se obtuvieron las respuestas en frecuencia del altavoz sin pantalla y con ella a diferentes distancias del altavoz. Posteriormente, los datos se analizaron con MatLab donde se calculó la atenuación, el factor de transmisión de la presión (PTF) y el análisis cepstrum. Finalmente, se realizó un modelo teórico del comportamiento de las pantallas perforadas basado en las placas perforadas utilizadas para atenuar el sonido entre distintas habitaciones. Como conclusión se llegó a que las pantallas curvadas son acústicamente más transparentes que las pantallas perforadas que a partir de 6 kHz son más acústicamente opacas. En las pantallas perforadas la atenuación depende del número de perforaciones por unidad de área y el diámetro de éstas. Dicha atenuación se reducirá si se reduce el diámetro de las perforaciones de la pantalla, o si se incrementa la cantidad de perforaciones. Acerca del efecto filtro peine, para obtener la mínima amplitud de éste la pantalla se deberá situar a una distancia entre 15 y 30 cm del altavoz, encontrando a la distancia de 30 cm que la última reflexión analizada a través de Cepstrum llega 5 ms más tarde que la señal directa, por lo cual no debería dañar el sonido ni la claridad del habla.

Low-resource language recognition using a fusion of phoneme posteriorgram counts, acoustic and glottal-based i-vectors

This paper presents a description of our system for the Albayzin 2012 LRE competition. One of the main characteristics of this evaluation was the reduced number of available files for training the system, especially for the empty condition where no training data set was provided but only a development set. In addition, the whole database was created from online videos and around one third of the training data was labeled as noisy files. Our primary system was the fusion of three different i-vector based systems: one acoustic system based on MFCCs, a phonotactic system using trigrams of phone-posteriorgram counts, and another acoustic system based on RPLPs that improved robustness against noise. A contrastive system that included new features based on the glottal source was also presented. Official and postevaluation results for all the conditions using the proposed metrics for the evaluation and the Cavg metric are presented in the paper.

Signal analysis and feature generation for pattern identification of partial discharges in high-voltage equipment

This paper proposes a method for the identification of different partial discharges (PDs) sources through the analysis of a collection of PD signals acquired with a PD measurement system. This method, robust and sensitive enough to cope with noisy data and external interferences, combines the characterization of each signal from the collection, with a clustering procedure, the CLARA algorithm. Several features are proposed for the characterization of the signals, being the wavelet variances, the frequency estimated with the Prony method, and the energy, the most relevant for the performance of the clustering procedure. The result of the unsupervised classification is a set of clusters each containing those signals which are more similar to each other than to those in other clusters. The analysis of the classification results permits both the identification of different PD sources and the discrimination between original PD signals, reflections, noise and external interferences. The methods and graphical tools detailed in this paper have been coded and published as a contributed package of the R environment under a GNU/GPL license.

Sea Bottom Classification by Means of Bathymetric LIDAR Data

Light Detection and Ranging (LIDAR) provides high horizontal and vertical resolution of spatial data located in point cloud images, and is increasingly being used in a number of applications and disciplines, which have concentrated on the exploit and manipulation of the data using mainly its three dimensional nature. Bathymetric LIDAR systems and data are mainly focused to map depths in shallow and clear waters with a high degree of accuracy. Additionally, the backscattering produced by the different materials distributed over the bottom surface causes that the returned intensity signal contains important information about the reflection properties of these materials. Processing conveniently these values using a Simplified Radiative Transfer Model, allows the identification of different sea bottom types. This paper presents an original method for the classification of sea bottom by means of information processing extracted from the images generated through LIDAR data. The results are validated using a vector database containing benthic information derived by marine surveys.

Extended phone log-likelihood ratio features and acoustic-based I-vectors for language recognition

This paper presents new techniques with relevant improvements added to the primary system presented by our group to the Albayzin 2012 LRE competition, where the use of any additional corpora for training or optimizing the models was forbidden. In this work, we present the incorporation of an additional phonotactic subsystem based on the use of phone log-likelihood ratio features (PLLR) extracted from different phonotactic recognizers that contributes to improve the accuracy of the system in a 21.4% in terms of Cavg (we also present results for the official metric during the evaluation, Fact). We will present how using these features at the phone state level provides significant improvements, when used together with dimensionality reduction techniques, especially PCA. We have also experimented with applying alternative SDC-like configurations on these PLLR features with additional improvements. Also, we will describe some modifications to the MFCC-based acoustic i-vector system which have also contributed to additional improvements. The final fused system outperformed the baseline in 27.4% in Cavg.

Some applications of distributed signal processing

In this work we review some earlier distributed algorithms developed by the authors and collaborators, which are based on two different approaches, namely, distributed moment estimation and distributed stochastic approximations. We show applications of these algorithms on image compression, linear classification and stochastic optimal control. In all cases, the benefit of cooperation is clear: even when the nodes have access to small portions of the data, by exchanging their estimates, they achieve the same performance as that of a centralized architecture, which would gather all the data from all the nodes.

Cortical auditory signal processing in poor readers

Magnetoencephalographic responses recorded from auditory cortex evoked by brief and rapidly successive stimuli differed between adults with poor vs. good reading abilities in four important ways. First, the response amplitude evoked by short-duration acoustic stimuli was stronger in the post-stimulus time range of 150–200 ms in poor readers than in normal readers. Second, response amplitude to rapidly successive and brief stimuli that were identical or that differed significantly in frequency were substantially weaker in poor readers compared with controls, for interstimulus intervals of 100 or 200 ms, but not for an interstimulus interval of 500 ms. Third, this neurological deficit closely paralleled subjects’ ability to distinguish between and to reconstruct the order of presentation of those stimulus sequences. Fourth, the average distributed response coherence evoked by rapidly successive stimuli was significantly weaker in the β- and γ-band frequency ranges (20–60 Hz) in poor readers, compared with controls. These results provide direct electrophysiological evidence supporting the hypothesis that reading disabilities are correlated with the abnormal neural representation of brief and rapidly successive sensory inputs, manifested in this study at the entry level of the cortical auditory/aural speech representational system(s).

Development of standards and tests for acceptance of traffic signal lenses. Final report.

Federal Highway Administration, Washington, D.C.

Suitability of acoustic perturbation measures in analysing periodic and nearly periodic voice signals

In recent years, acoustic perturbation measurement has gained clinical and research popularity due to the ease of availability of commercial acoustic analysing software packages in the market. However, because the measurement itself depends critically on the accuracy of frequency tracking from the voice signal, researchers argue that perturbation measures are not suitable for analysing dysphonic voice samples, which are aperiodic in nature. This study compares the fundamental frequency, relative amplitude perturbation, shimmer percent and noise-to-harmonic ratio between a group of dysphonic and non-dysphonic subjects. One hundred and twelve dysphonic subjects ( 93 females and 19 males) and 41 non-dysphonic subjects ( 35 females and 6 males) participated in the study. All the 153 voice samples were categorized into type I ( periodic or nearly periodic), type II ( signals with subharmonic frequencies that approach the fundamental frequency) and type III ( aperiodic) signals. Only the type I ( periodic and nearly periodic) voice signals were acoustically analysed for perturbation measures. Results revealed that the dysphonic female group presented significantly lower fundamental frequency, significantly higher relative amplitude perturbation and shimmer percent values than the non-dysphonic female group. However, none of these three perturbation measures were able to differentiate between male dysphonic and male non-dysphonic subjects. The noise-to-harmonic ratio failed to differentiate between the dysphonic and non-dysphonic voices for both gender groups. These results question the sensitivity of acoustic perturbation measures in detecting dysphonia and suggest that contemporary acoustic perturbation measures are not suitable for analysing dysphonic voice signals, which are even nearly periodic. Copyright (C) 2005 S. Karger AG, Basel.

Habitat type and density influence vocal signal design in satin bowerbirds

This study provided a thorough test of the acoustic adaptation hypothesis using a within-species comparison of call structure involving a wide range of habitat types, an objective measure of habitat density and direct measures of habitat-related attenuation. The structure of the bower advertisement call of the satin bowerbird was measured in 16 populations from throughout the species' range and related to the habitat type and density at each site. Transmission of white noise, pure tones and different bowerbird dialects was measured in five of six habitat types inhabited by satin bowerbirds. Bowerbird advertisement call structure converged in similar habitats but diverged among different habitats; this pattern was apparent at both continent-wide and local geographical scales. Bowerbirds' call structures differed with changes in habitat density, consistent with the acoustic adaptation hypothesis. Lower frequencies and less frequency modulation were utilized in denser habitats such as rainforest and higher frequencies and more frequency modulation were used in the more open eucalypt-dominated habitats. The white noise and pure tone transmission measurements indicated that different habitats varied in their sound transmission properties in a manner consistent with the observed variation in satin bowerbird vocalizations. There was no effect of geographical proximity of recording locations, nor was there the predicted inverse relationship between frequency and body size. These findings indicate that the transmission qualities of different habitats have had a major influence on variation in vocal phenotypes in this species. In addition, previously published molecular data for this species suggest that there is no effect of genetic relatedness on call similarity among satin bowerbird populations.

Signal processing in a loudspeaking telephone

One of the major problems associated with communication via a loudspeaking telephone (LST) is that, using analogue processing, duplex transmission is limited to low-loss lines and produces a low acoustic output. An architectural for an instrument has been developed and tested, which uses digital signal processing to provide duplex transmission between a LST and a telopnone handset over most of the B.T. network. Digital adaptive-filters are used in the duplex LST to cancel coupling between the loudspeaker and microphone, and across the transmit to receive paths of the 2-to-4-wire converter. Normal movement of a person in the acoustic path causes a loss of stability by increasing the level of coupling from the loudspeaker to the microphone, since there is a lag associated the adaptive filters learning about a non-stationary path, Control of the loop stability and the level of sidetone heard by the hadset user is by a microprocessoe, which continually monitors the system and regulates the gain. The result is a system which offers the best compromise available based on a set of measured parameters.A theory has been developed which gives the loop stability requirements based on the error between the parameters of the filter and those of the unknown path. The programme to develope a low-cost adaptive filter in LST produced a low-cost adaptive filter in LST produced a unique architecture which has a number of features not available in any similar system. These include automatic compensation for the rate of adaptation over a 36 dB range of output level, , 4 rates of adaptation (with a maximum of 465 dB/s), plus the ability to cascade up to 4 filters without loss o performance. A complex story has been developed to determine the adptation which can be achieved using finite-precision arithmatic. This enabled the development of an architecture which distributed the normalisation required to achieve optimum rate of adaptation over the useful input range. Comparison of theory and measurement for the adaptive filter show very close agreement. A single experimental LST was built and tested on connections to hanset telephones over the BT network. The LST demonstrated that duplex transmission was feasible using signal processing and produced a more comfortable means of communication beween people than methods emplying deep voice-switching to regulate the local-loop gain. Although, with the current level of processing power, it is not a panacea and attention must be directed toward the physical acoustic isolation between loudspeaker and microphone.

Task classification and decision processes in monitoring behaviour

Task classification is introduced as a method for the evaluation of monitoring behaviour in different task situations. On the basis of an analysis of different monitoring tasks, a task classification system comprising four task 'dimensions' is proposed. The perceptual speed and flexibility of closure categories, which are identified with signal discrimination type, comprise the principal dimension in this taxonomy, the others being sense modality, the time course of events, and source complexity. It is also proposed that decision theory provides the most complete method for the analysis of performance in monitoring tasks. Several different aspects of decision theory in relation to monitoring behaviour are described. A method is also outlined whereby both accuracy and latency measures of performance may be analysed within the same decision theory framework. Eight experiments and an organizational study are reported. The results show that a distinction can be made between the perceptual efficiency (sensitivity) of a monitor and his criterial level of response, and that in most monitoring situations, there is no decrement in efficiency over the work period, but an increase in the strictness of the response criterion. The range of tasks exhibiting either or both of these performance trends can be specified within the task classification system. In particular, it is shown that a sensitivity decrement is only obtained for 'speed' tasks with a high stimulation rate. A distinctive feature of 'speed' tasks is that target detection requires the discrimination of a change in a stimulus relative to preceding stimuli, whereas in 'closure' tasks, the information required for the discrimination of targets is presented at the same point In time. In the final study, the specification of tasks yielding sensitivity decrements is shown to be consistent with a task classification analysis of the monitoring literature. It is also demonstrated that the signal type dimension has a major influence on the consistency of individual differences in performance in different tasks. The results provide an empirical validation for the 'speed' and 'closure' categories, and suggest that individual differences are not completely task specific but are dependent on the demands common to different tasks. Task classification is therefore shovn to enable improved generalizations to be made of the factors affecting 1) performance trends over time, and 2) the consistencv of performance in different tasks. A decision theory analysis of response latencies is shown to support the view that criterion shifts are obtained in some tasks, while sensitivity shifts are obtained in others. The results of a psychophysiological study also suggest that evoked potential latency measures may provide temporal correlates of criterion shifts in monitoring tasks. Among other results, the finding that the latencies of negative responses do not increase over time is taken to invalidate arousal-based theories of performance trends over a work period. An interpretation in terms of expectancy, however, provides a more reliable explanation of criterion shifts. Although the mechanisms underlying the sensitivity decrement are not completely clear, the results rule out 'unitary' theories such as observing response and coupling theory. It is suggested that an interpretation in terms of the memory data limitations on information processing provides the most parsimonious explanation of all the results in the literature relating to sensitivity decrement. Task classification therefore enables the refinement and selection of theories of monitoring behaviour in terms of their reliability in generalizing predictions to a wide range of tasks. It is thus concluded that task classification and decision theory provide a reliable basis for the assessment and analysis of monitoring behaviour in different task situations.