Model-based approaches to handling additive noise in reverberant environments

Model-based approaches to handle additive and convolutional noise have been extensively investigated and used. However, the application of these schemes to handling reverberant noise has received less attention. This paper examines the extension of two standard additive/convolutional noise approaches to handling reverberant noise. The first is an extension of vector Taylor series (VTS) compensation, reverberant VTS, where a mismatch function including reverberant noise is used. The second scheme modifies constrained MLLR to allow a wide-span of frames to be taken into account and projected into the required dimensionality. To allow additive noise to be handled, both these schemes are combined with standard VTS. The approaches are evaluated and compared on two tasks, MC-WSJ-AV, and a reverberant simulated version of AURORA-4. © 2011 IEEE.

Investigating landing gear noise using fly-over data: The case of a Boeing 747-400

Modeling the noise originating from a landing gear has proven to be a challenging task, because of its complicated structure. In full-scale, landing gear noise can only be investigated experimentally by source localization techniques and fly-over measurements with microphone arrays. In the present work, measurements of a Boeing B747-400 were used to determine the contribution of the landing gear to the overall noise emitted during a fly-over and how the broadband noise from the landing gear scales with the flight velocity. A tonal source from the nose landing gear was identified at 380 Hz with a harmonic at 760 Hz and it most likely originates from a cavity. It was also found that the Power Spectral Density (PSD) of the high frequency broadband component varies linearly with frequency and there is some scaling with the ow velocity. Finally, the nose landing gear was shown to be a significant contributor to the overall airframe noise as expected.

Experimental study of surface roughness noise

A turbulent boundary-layer flow over a rough wall generates a dipole sound field as the near-field hydrodynamic disturbances in the turbulent boundary-layer scatter into radiated sound at small surface irregularities. In this paper, phased microphone arrays are applied to the experimental study of surface roughness noise. The radiated sound from two rough plates and one smooth plate in an open jet is measured at three streamwise locations, and the beamforming source maps demonstrate the dipole directivity. Higher source strengths can be observed in the rough plates than the smooth plate, and the rough plates also enhance the trailing-edge noise. A prediction scheme in previous theoretical work is used to describe the strength of a distribution of incoherent dipoles over the rigid plate and to simulate the sound detected by the microphone array. Source maps of measurement and simulation exhibit encouraging similarities in both source pattern and source strength, which confirms the dipole nature and the predicted magnitude of roughness noise. The simulations underestimate the streamwise gradient of the source strengths and overestimate the source strengths at the highest frequency. © 2007 by Yu Liu and Ann P. Dowling.

Aeroacoustic source measurement methods for characterizing the sound generated by ducted flow devices with higher-order modes.

The International Organization for Standardization (ISO) method 5136 is widely used in industry and academia to determine the sound power radiated into a duct by fans and other flow devices. The method involves placing the device at the center of a long cylindrical duct with anechoic terminations at each end to eliminate reflections. A single off-axis microphone is used on the inlet and outlet sides that can theoretically capture the plane-wave mode amplitudes but this does not provide enough information to fully account for higher-order modes. In this study, the "two-port" source model is formulated to include higher-order modes and applied for the first three modes. This requires six independent surface pressure measurements on each side or "port." The resulting experimental set-up is much shorter than the ISO rig and does not require anechoic terminations. An array of six external loudspeaker sources is used to characterize the passive part of the two-port model and the set-up provides a framework to account for transmission of higher-order modes through a fan. The relative importance of the higher-order modes has been considered and their effect on inaccuracies when using the ISO method to find source sound power has been analyzed.

Investigation on a pressure-gradient fiber laser hydrophone

In this paper, a pressure-gradient fiber laser hydrophone is demonstrated. Two brass diaphragms are installed at the end of a metal cylinder as sensing elements. A distributed feedback fiber laser, fixed at the center of the two diaphragms, is elongated or shortened due to the acoustic wave. There are two orifices at the middle of the cylinder. So this structure can work as a pressure-gradient microphone in the acoustic field. Furthermore, the hydrostatic pressure is self-compensated and an ultra-thin dimension is achieved. Theoretical analysis is given based on the electro-acoustic theory. Field trials are carried out to test the performance of the hydrophone. A sensitivity of 100 nm MPa-1 has been achieved. Due to the small dimensions, no directivity is found in the test.

Towards Real-time Speech Emotion Recognition for Affective E-Learning

The original article is available as an open access file on the Springer website in the following link: http://link.springer.com/article/10.1007/s10639-015-9388-2

Wideband measurement of the acoustic impedance of tubular objects

A method is discussed for measuring the acoustic impedance of tubular objects that gives accurate results for a wide range of frequencies. The apparatus that is employed is similar to that used in many previously developed methods; it consists of a cylindrical measurement duct fitted with several microphones, of which two are active in each measurement session, and a driver at one of its ends. The object under study is fitted at the other end. The impedance of the object is determined from the microphone signals obtained during excitation of the air inside the 1 duct by the driver, and from three coefficients that are pre-determined using four calibration measurements with closed cylindrical tubes. The calibration procedure is based on the simple mathematical relationships between the impedances of the calibration tubes, and does not require knowledge of the propagation constant. Measurements with a cylindrical tube yield an estimate of the attenuation constant for plane waves, which is found to differ from the theoretical prediction by less than 1.4% in the frequency range 1 kHz-20 kHz. Impedance measurements of objects with abrupt changes in diameter are found to be in good agreement with multimodal theory.

Time domain wave separation using multiple microphones

Methods of measuring the acoustic behavior of tubular systems can be broadly characterized as steady state measurements, where the measured signals are analyzed in terms of infinite duration sinusoids, and reflectometry measurements which exploit causality to separate the forward and backward going waves in a duct. This paper sets out a multiple microphone reflectometry technique which performs wave separation by using time domain convolution to track the forward and backward going waves in a cylindrical source tube. The current work uses two calibration runs (one for forward going waves and one for backward going waves) to measure the time domain transfer functions for each pair of microphones. These time domain transfer functions encode the time delay, frequency dependent losses and microphone gain ratios for travel between microphones. This approach is applied to the measurement of wave separation, bore profile and input impedance. The work differs from existing frequency domain methods in that it combines the information of multiple microphones within a time domain algorithm, and differs from existing time domain methods in its inclusion of the effect of losses and gain ratios in intermicrophone transfer functions.

A Frequency Domain Algorithm for Wave Separation

We propose a frequency domain adaptive algorithm for
wave separation in wind instruments. Forward and backward travelling waves are obtained from the signals acquired by two microphones placed along the tube, while the
separation ?lter is adapted from the information given by a
third microphone. Working in the frequency domain has a
series of advantages, among which are the ease of design of
the propagation ?lter and its differentiation with respect to
its parameters.
Although the adaptive algorithm was developed as a ?rst
step for the estimation of playing parameters in wind instruments it can also be used, without any modi?cations, for
other applications such as in-air direction of arrival (DOA)
estimation. Preliminary results on these applications will
also be presented.

Investigation on localisation accuracy for first and higher order ambisonics reproduced sound sources

Ambisonics and higher order ambisonics (HOA) technologies aim at reproducing sound field either synthesised or previously recorded with dedicated microphones. Based on a spherical harmonic decomposition, the sound field is more precisely described when higher-order components are used. The presented study evaluated the perceptual and objective localisation accuracy of the sound field encoded with four microphones of order one to four and decoded over a ring of loudspeakers. A perceptual test showed an improvement of the localisation with higher order ambisonic microphones. Reproduced localisation indices were estimated for the four microphones and the respective synthetic systems of order one to four. The perceptual and objective analysis revealed the same conclusions. The localisation accuracy depends on the ambisonic order as well as the source incidence. Furthermore, impairments linked to the microphones were highlighted.

Binaural Reproduction of Finite Difference Simulations Using Spherical Array Processing

Due to its efficiency and simplicity, the finite-difference time-domain method is becoming a popular choice for solving wideband, transient problems in various fields of acoustics. So far, the issue of extracting a binaural response from finite difference simulations has only been discussed in the context of embedding a listener geometry in the grid. In this paper, we propose and study a method for binaural response rendering based on a spatial decomposition of the sound field. The finite difference grid is locally sampled using a volumetric array of receivers, from which a plane wave density function is computed and integrated with free-field head related transfer functions, in the spherical harmonics domain. The volumetric array is studied in terms of numerical robustness and spatial aliasing. Analytic formulas that predict the performance of the array are developed, facilitating spatial resolution analysis and numerical binaural response analysis for a number of finite difference schemes. Particular emphasis is placed on the effects of numerical dispersion on array processing and on the resulting binaural responses. Our method is compared to a binaural simulation based on the image method. Results indicate good spatial and temporal agreement between the two methods.

Ressonância magnética em estudos de produção de fala

Estudar os mecanismos subjacentes à produção de fala é uma tarefa complexa e exigente, requerendo a obtenção de dados mediante a utilização de variadas técnicas, onde se incluem algumas modalidades imagiológicas. De entre estas, a Ressonância Magnética (RM) tem ganho algum destaque, nos últimos anos, posicionando-se como uma das mais promissoras no domínio da produção de fala. Um importante contributo deste trabalho prende-se com a otimização e implementação de protocolos (RM) e proposta de estratégias de processamento de imagem ajustados aos requisitos da produção de fala, em geral, e às especificidades dos diferentes sons. Para além disso, motivados pela escassez de dados para o Português Europeu (PE), constitui-se como objetivo a obtenção de dados articulatórios que permitam complementar informação já existente e clarificar algumas questões relativas à produção dos sons do PE (nomeadamente, consoantes laterais e vogais nasais). Assim, para as consoantes laterais foram obtidas imagens RM (2D e 3D), através de produções sustidas, com recurso a uma sequência Eco de Gradiente (EG) rápida (3D VIBE), no plano sagital, englobando todo o trato vocal. O corpus, adquirido por sete falantes, contemplou diferentes posições silábicas e contextos vocálicos. Para as vogais nasais, foram adquiridas, em três falantes, imagens em tempo real com uma sequência EG - Spoiled (TurboFLASH), nos planos sagital e coronal, obtendo-se uma resolução temporal de 72 ms (14 frames/s). Foi efetuada aquisição sincronizada das imagens com o sinal acústico mediante utilização de um microfone ótico. Para o processamento e análise de imagem foram utilizados vários algoritmos semiautomáticos. O tratamento e análise dos dados permitiu efetuar uma descrição articulatória das consoantes laterais, ancorada em dados qualitativos (e.g., visualizações 3D, comparação de contornos) e quantitativos que incluem áreas, funções de área do trato vocal, extensão e área das passagens laterais, avaliação de efeitos contextuais e posicionais, etc. No que respeita à velarização da lateral alveolar /l/, os resultados apontam para um /l/ velarizado independentemente da sua posição silábica. Relativamente ao /L/, em relação ao qual a informação disponível era escassa, foi possível verificar que a sua articulação é bastante mais anteriorizada do que tradicionalmente descrito e também mais extensa do que a da lateral alveolar. A resolução temporal de 72 ms conseguida com as aquisições de RM em tempo real, revelou-se adequada para o estudo das características dinâmicas das vogais nasais, nomeadamente, aspetos como a duração do gesto velar, gesto oral, coordenação entre gestos, etc. complementando e corroborando resultados, já existentes para o PE, obtidos com recurso a outras técnicas instrumentais. Para além disso, foram obtidos novos dados de produção relevantes para melhor compreensão da nasalidade (variação área nasal/oral no tempo, proporção nasal/oral). Neste estudo, fica patente a versatilidade e potencial da RM para o estudo da produção de fala, com contributos claros e importantes para um melhor conhecimento da articulação do Português, para a evolução de modelos de síntese de voz, de base articulatória, e para aplicação futura em áreas mais clínicas (e.g., perturbações da fala).

An FPGA based decimation filter processor design for real-time continuous-time Σ−Δ modulator performance measurement and evaluation

This paper reports on a Field Programmable Gate Array (FPGA) implementation as well as prototyping for real-time testing of a low complexity high efficiency decimation filter processor which is deployed in conjunction with a custom built low-power jitter insensitive Continuous Time (CT) Sigma-Delta (Σ-Δ) Modulator to measure and assess its performance. The CT Σ-Δ modulator/decimation filter cascade can be used in integrated all-digital microphone interfaces for a variety of applications including mobile phone handsets, wireless handsets as well as other applications requiring all-digital microphones. The work reported here concentrates on the design and implementation as well as prototyping on a Xilinx Spartan 3 FPGA development system and real-time testing of the decimation processing part deploying All-Pass based structures to process the bit stream coming from CT Σ-Δ modulator hence measuring in real-time and fully assessing the modulator's performance.

Recollections of a conference interpreter: a brief encounter with the European Parliament

Polissema: Revista de Letras do ISCAP 2001/N.º 1- Tradução

Smart object for 3D interaction

This paper reports on the creation of an interface for 3D virtual environments, computer-aided design applications or computer games. Standard computer interfaces are bound to 2D surfaces, e.g., computer mouses, keyboards, touch pads or touch screens. The Smart Object is intended to provide the user with a 3D interface by using sensors that register movement (inertial measurement unit), touch (touch screen) and voice (microphone). The design and development process as well as the tests and results are presented in this paper. The Smart Object was developed by a team of four third-year engineering students from diverse scientific backgrounds and nationalities during one semester.