Compensation des non-linéarités des systèmes haut-parleurs à pavillon

The assumptions of linearity and autonomy assumed in circuit theory are known to be respected only partially by electroacoustic transducers. In particular, the dynamics range of the latter is limited by the audible nonlinear effects that they are causing. In the domain of active noise control, the distortion products limit noise reduction performance. The present thesis work stems from characterization of these observations. We want to design a real-time system compensating the nonlinearities of electroacoustic devices. The first part of this report concerns the nonlinearity effects. It is shown that the use of classical characterization methods of an electroacoustic device (harmonic distortion, intermodulation, frequency difference ...) is limited. Hence, they can not be used to determine the nonlinearity laws themselves, but rather only their effects for arbitrary excitations. First, a method based on multitone harmonic excitations has been proposed and validated. It has been applied for the characterization of loudspeaker prototypes using several different technologies, which are used for active noise reduction in an aircraft turboreactor. The second part of this work addresses the elaboration and validation of a nonlinearity effect compensation method. This method is based on the description of the nonlinearities by the Volterra series. For this, we need to place an upstream system, characterized by the inverse nonlinearity law of the loudspeaker system. After having determined precisely the Volterra kernels of the system to be characterized, the kernels of the upstream compensation system are determined by assuming that their cascade obeys to a linear law. For this purpose, a kernel measurement method in the frequency domain has been developed, validated and tested. Since we are concerned with loudspeakers, we took their flying time into account. The compensation method has been validated by computing the simulated response of the compensation circuit. Resulting analog signals have then been applied to the loudspeaker. Measured performances fulfill the expectations. In the last part, the compensation method has been applied in a real-time Digital Signal Processing) DSP controller, which allowed to realize a demonstrator to the implemented for future industrial applications.

Differential microphone array for speech recognition

Hervé Lissek, Philippe Martin

Speech recognition applications embedded on a PDA are already available on the market. The usual hardware for this kind of systems is a single microphone mounted on the PDA, giving good results within quiet environments. Though, the recognition rate falls drastically as the signal to noise ratio decreases. Arrays of microphones are then particularly interesting, allowing the discrimination of useful sounds sources within parasitic ones, thanks to an improved directivity pattern. In speech recognition applications on PDA, a trade-off is to be found on an array small enough to guarantee the ergonomics of the whole system, while operating on a bandwidth covering the frequency range of human voice, from 300 Hz up to 6 kHz and low distortion ratio. Differential arrays are well adapted to fulfill these specifications, since they are known to be robust and allowing small dimensions for a high directivity index. The performances could be improved by the addition of adaptive post-filtering and noise reduction algorithms. This work describes the design and the implementation of a recording device dedicated to speech recognition applications on PDA, based on overlapping differential arrays. The assessment of its performances in noisy environment are carried out and show the system efficiency.

2008

Analysis and Design of Ultra-Wideband Antennas in the Spectral and Temporal Domains

Gabriela Quintero Diaz De Leon

Formal research on Ultra-wideband (UWB) technology started in the U.S. in 2002; where its definition was formulated and the frequency allocation in that country established. However, the first ultra-wideband transmission was done during the confirmation of Maxwell's equations by Heinrich Hertz in 1886, the first ever reported radio transmission. Sparks or very short pulses were sent and received at a distance away from its source. UWB antennas have existed since those times, but technology has been mostly focused on the transmission of continuous wave narrowband signals. The largest problematic encountered in today's UWB antennas is that characterization methods were, since the beginning, meant to describe narrowband antennas. The frequency characteristics of these antennas are constant over their operational bandwidth, hence they can be fully characterized in the frequency domain. UWB antennas, on the other hand, are meant to transmit pulsed signals. Analyzing them only in the frequency domain is not enough to fully evaluate their performance, as pulse distortion is an important parameter that should be controlled. An extended literature review is included in the first part of this thesis, where the available UWB antenna characterization methods are evaluated. Many limitations were encountered, showing that the available techniques do not analyze simultaneously the most important parameters of an UWB antenna. A new characterization method is proposed: the System Fidelity Factor (SFF). Its main purpose is to incorporate frequency, time and space characteristics of a two-antennas system to compare UWB antennas in an efficient way. This is achieved with the correlation between the received and the input pulses, quantifying the distortion produced by the system. The SFF is an interesting tool because both simulations and measurements can be done in a simple and straight-forward manner, using tools that are commonly available in any antenna lab. Any combination of antennas can be analyzed and the free-space channel can be easily replaced with any other environment. Several UWB antennas were designed and used to prove the efficiency of the SFF. A good agreement between the proposed simulation techniques and the measurements was achieved. The antennas are novel designs specially developed for UWB applications under the American or European regulations. The last part of the thesis presents these antennas and analyzes them using the SFF to compare their performance.

EPFL2010

Endoscopic low coherence interferometry applied to tri-dimensional contouring of upper airways

Yves Delacrétaz

This thesis presents a novel approach for optical tri-dimensional contouring, requiring only one acquisition to extract a contour depth. It is based on an interferometric setup, using a reduced coherence light source, and is fully adaptable to endoscopic procedure commonly in used in ear-nose-throat (ENT) medicine. It has been demonstrated that the method can be successfully applied to imaging of porcine upper airways. Although this method uses interferometry to obtain depth range measurement, the phase signal is not directly evaluated, rather the location of the coherent superposition of two waves is extracted, and thus provides a robust technique even in harsh environment. The key feature is to extract the coherent area on the image by locating intentionally induced fringes created by off-axis superposition of a smooth reference wave and an object wave coming from a rough or diffusing sample through an optical system with limited aperture. A model for the propagation of the light through the optical system has been developed. It takes into account the broadened emission spectrum of the light source, as well as the statistical speckle effect induced by the illumination of a rough surface with partially coherent light. Simulated results have been carefully compared with measurements, both in the spatial domain and in the Fourier domain (spatial frequencies response), which has permitted to better understand the processes involved in the creation of the interference fringes, and improve the design of the device. Different filters have been investigated in order to extract the depth information from acquired interferograms. By using Gabor filterbanks, which do not need any a priori knowledge of the signal, selectivity of the extracted signal has been enhanced by a factor of 1.3 compared to a simple local Fourier spectrum evaluation, and time of the extraction procedure divided by 6, even when compared to fast iterative calculation in the Fourier spectrum. Moreover, by imposing the frequency and/or orientation of the signal to detect, which can be known by a simple calibration procedure, it has been proved that the selectivity can be further enhanced by a factor of 2.5, keeping the calculation time as short or even shorter. A compact prototype has been designed and built, with a specific multiple fibers arrangement for illumination. The system is separated in two parts: the first one is directly attached to the rigid endoscope, and contains imaging as well as recombining object/reference wave optics, and the detector. It fits into a box with overall dimensions of 115 × 83 × 94 mm. The second part contains the laser source, the scanning arm and the injection optics for the fibered system. It is designed to be left on a small cart. This device can potentially be brought to the bedside, and used during routine endoscopic check-up. Associated with the device, a complete framework has been elaborated in order to simplify the acquisition procedure as well as the signal processing. This results in a fully automated environment taking in charge hardware control, data storage and signal processing, up to the tri-dimensional scene rendering.