918 resultados para Auditory Warning Signals.
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Fine magnetic particles (sizeffi100A ˚ ) belonging to the series ZnxFe1 xFe2O4 were synthesized by cold co-precipitation methods and their structural properties were evaluated using X-ray diffraction. Magnetization studies have been carried out using vibrating sample magnetometry (VSM) showing near-zero loss loop characteristics. Ferrofluids were then prepared employing these fine magnetic powders using oleic acid as surfactant and kerosene as carrier liquid by modifying the usually reported synthesis technique in order to induce anisotropy and enhance the magneto-optical signals. Liquid thin films of these fluids were prepared and field-induced laser transmission through these films was studied. The transmitted light intensity decreases at the centre with applied magnetic field in a linear fashion when subjected to low magnetic fields and saturate at higher fields. This is in accordance with the saturation in cluster formation. The pattern exhibited by these films in the presence of different magnetic fields was observed with the help of a CCD camera and was recorded photographically
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Modeling nonlinear systems using Volterra series is a century old method but practical realizations were hampered by inadequate hardware to handle the increased computational complexity stemming from its use. But interest is renewed recently, in designing and implementing filters which can model much of the polynomial nonlinearities inherent in practical systems. The key advantage in resorting to Volterra power series for this purpose is that nonlinear filters so designed can be made to work in parallel with the existing LTI systems, yielding improved performance. This paper describes the inclusion of a quadratic predictor (with nonlinearity order 2) with a linear predictor in an analog source coding system. Analog coding schemes generally ignore the source generation mechanisms but focuses on high fidelity reconstruction at the receiver. The widely used method of differential pnlse code modulation (DPCM) for speech transmission uses a linear predictor to estimate the next possible value of the input speech signal. But this linear system do not account for the inherent nonlinearities in speech signals arising out of multiple reflections in the vocal tract. So a quadratic predictor is designed and implemented in parallel with the linear predictor to yield improved mean square error performance. The augmented speech coder is tested on speech signals transmitted over an additive white gaussian noise (AWGN) channel.
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Speech is the primary, most prominent and convenient means of communication in audible language. Through speech, people can express their thoughts, feelings or perceptions by the articulation of words. Human speech is a complex signal which is non stationary in nature. It consists of immensely rich information about the words spoken, accent, attitude of the speaker, expression, intention, sex, emotion as well as style. The main objective of Automatic Speech Recognition (ASR) is to identify whatever people speak by means of computer algorithms. This enables people to communicate with a computer in a natural spoken language. Automatic recognition of speech by machines has been one of the most exciting, significant and challenging areas of research in the field of signal processing over the past five to six decades. Despite the developments and intensive research done in this area, the performance of ASR is still lower than that of speech recognition by humans and is yet to achieve a completely reliable performance level. The main objective of this thesis is to develop an efficient speech recognition system for recognising speaker independent isolated words in Malayalam.
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Presently different audio watermarking methods are available; most of them inclined towards copyright protection and copy protection. This is the key motive for the notion to develop a speaker verification scheme that guar- antees non-repudiation services and the thesis is its outcome. The research presented in this thesis scrutinizes the field of audio water- marking and the outcome is a speaker verification scheme that is proficient in addressing issues allied to non-repudiation to a great extent. This work aimed in developing novel audio watermarking schemes utilizing the fun- damental ideas of Fast-Fourier Transform (FFT) or Fast Walsh-Hadamard Transform (FWHT). The Mel-Frequency Cepstral Coefficients (MFCC) the best parametric representation of the acoustic signals along with few other key acoustic characteristics is employed in crafting of new schemes. The au- dio watermark created is entirely dependent to the acoustic features, hence named as FeatureMark and is crucial in this work. In any watermarking scheme, the quality of the extracted watermark de- pends exclusively on the pre-processing action and in this work framing and windowing techniques are involved. The theme non-repudiation provides immense significance in the audio watermarking schemes proposed in this work. Modification of the signal spectrum is achieved in a variety of ways by selecting appropriate FFT/FWHT coefficients and the watermarking schemes were evaluated for imperceptibility, robustness and capacity char- acteristics. The proposed schemes are unequivocally effective in terms of maintaining the sound quality, retrieving the embedded FeatureMark and in terms of the capacity to hold the mark bits. Robust nature of these marking schemes is achieved with the help of syn- chronization codes such as Barker Code with FFT based FeatureMarking scheme and Walsh Code with FWHT based FeatureMarking scheme. An- other important feature associated with this scheme is the employment of an encryption scheme towards the preparation of its FeatureMark that scrambles the signal features that helps to keep the signal features unreve- laed. A comparative study with the existing watermarking schemes and the ex- periments to evaluate imperceptibility, robustness and capacity tests guar- antee that the proposed schemes can be baselined as efficient audio water- marking schemes. The four new digital audio watermarking algorithms in terms of their performance are remarkable thereby opening more opportu- nities for further research.
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Abstract: The paper describes an auditory interface using directional sound as a possible support for pilots during approach in an instrument landing scenario. Several ways of producing directional sounds are illustrated. One using speaker pairs and controlling power distribution between speakers is evaluated experimentally. Results show, that power alone is insufficient for positioning single isolated sound events, although discrimination in the horizontal plane performs better than in the vertical. Additional sound parameters to compensate for this are proposed.
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We discuss the problem of finding sparse representations of a class of signals. We formalize the problem and prove it is NP-complete both in the case of a single signal and that of multiple ones. Next we develop a simple approximation method to the problem and we show experimental results using artificially generated signals. Furthermore,we use our approximation method to find sparse representations of classes of real signals, specifically of images of pedestrians. We discuss the relation between our formulation of the sparsity problem and the problem of finding representations of objects that are compact and appropriate for detection and classification.
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In order to estimate the motion of an object, the visual system needs to combine multiple local measurements, each of which carries some degree of ambiguity. We present a model of motion perception whereby measurements from different image regions are combined according to a Bayesian estimator --- the estimated motion maximizes the posterior probability assuming a prior favoring slow and smooth velocities. In reviewing a large number of previously published phenomena we find that the Bayesian estimator predicts a wide range of psychophysical results. This suggests that the seemingly complex set of illusions arise from a single computational strategy that is optimal under reasonable assumptions.
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La audición es el segundo mecanismo sensorial más importante después de la visión para obtener información durante la operación de una aeronave. Les permite a los pilotos percibir, procesar identificar los sonidos del ambiente que los rodea. Así necesita oír bien tanto en vuelo como en tierra, especialmente entre 500 y 3000 Hz para la recepción del lenguaje hablado y de las señales auditivas. Objetivo: Determinar los cambios progresivos en el tiempo y las frecuencias auditivas que se afectan en las audiometrías de los pilotos militares de las fuerzas militares en los años 2009, 2010 y 2011. Material y Métodos: Se trata de un estudio longitudinal de cohorte en el cual se identificará el comportamiento de las audiometrías de la población de pilotos de las fuerzas militares de Colombia en los años 2009, 2010 y 2011. Se hará una revisión retrospectiva de dichas audiometrías. Para dicho fin se tomó la población de pilotos de fuerzas militares que fueron distribuidos en grupos de pilotos de aeronave de ala fija que corresponden a 47 pilotos y ala rotatoria que son 155. Conclusiones: Se encontró que la frecuencia mas alterada en la población total fue la de 6000 Hz, que en lo pilotos de ala fija las frecuencias más afectadas fueron las de 4000 Hz y la de 6000Hz, la frecuencia más afectada en los pilotos de ala rotatoria fueron las de 4000 Hz, 6000 Hz y 8000 Hz, con lo que se concluye que la exposición en los pilotos afecta las frecuencias altas en las audiometrías. Se observó una relación con el número de horas de vuelo y las alteraciones audiométricas encontrándose una alteración en los pilotos entre 1000 y 4000 horas de vuelo en las frecuencias de 4000 Hz, 6000 Hz y 8000 Hz y una alteración de las todas las frecuencias en aquellos pilotos con más de 5000 horas de vuelo en el año 2009, presentando posterior recuperación en los años posteriores sin poder determinar en este estudio las causas de dicha recuperación. Los pilotos de ala rotatoria presentaron un incremento sostenido en todas las frecuencias en comparación con los pilotos de ala fija.
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Resumen basado en el de la publicación. Resumen en español
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This paper reviews a study to determine if loss of speech discrimination is related to age and patients with audiograms showing steep high-frequency losses.
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This paper discusses a study conducted to test sound discrimination abilities of the chinchilla.
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This paper is a review of auditory training using amplification for hearing impaired children.
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This paper discusses auditory perception differences in aphasic and non-aphasic children
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The purpose of this project was to determine whether a monaural FM or binaural FM fitting strategy is more appropriate for children with Auditory Processing Disorders.
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This paper studies the auditory, visual and combined audio-visual recognition of vowels by severely and profoundly hearing impaired children.