Array signal processing for source localization and enhancement

“A common approach to the wide-band microphone array problem is to assume a certain array geometry and then design optimal weights (often in subbands) to meet a set of desired criteria. In addition to weights, we consider the geometry of the microphone arrangement to be part of the optimization problem. Our approach is to use particle swarm optimization (PSO) to search for the optimal geometry while using an optimal weight design to design the weights for each particle’s geometry. The resulting directivity indices (DI’s) and white noise SNR gains (WNG’s) form the basis of the PSO’s fitness function. Another important consideration in the optimal weight design are several regularization parameters. By including those parameters in the particles, we optimize their values as well in the operation of the PSO. The proposed method allows the user great flexibility in specifying desired DI’s and WNG’s over frequency by virtue of the PSO fitness function. Although the above method discusses beam and nulls steering for fixed locations, in real time scenarios, it requires us to estimate the source positions to steer the beam position adaptively. We also investigate source localization of sound and RF sources using machine learning techniques. As for the RF source localization, we consider radio frequency identification (RFID) antenna tags. Using a planar RFID antenna array with beam steering capability and using received signal strength indicator (RSSI) value captured for each beam position, the position of each RFID antenna tag is estimated. The proposed approach is also shown to perform well under various challenging scenarios”--Abstract, page iv

Speech Enhancement using Fiber Acoustic Sensor

With the development of IoT (Internet of Things) services and devices, the voice command becomes a more and more important tool for human computer interaction. However, the audio signal recorded by the conventional omni-directional microphone is easy to be corrupted by the environmental noise like interference speech. Although the conventional beamforming techniques are able to point the main lobe of beam pattern at the desired speaker, it requires several omni microphones to form a microphone array, which will occupy large space on an IoT device. Many researchers are devoting their efforts to inventing a microphone of small size that can create directional beam pattern. Recently, researchers get inspirations from the spider’s way to sense the acoustic wave. They invented a new small-size acoustic sensor made of spider silks. This acoustic sensor has a frequency-independent dipole beam pattern for wideband audio signal. Utilizing this fiber acoustic sensor, two compact microphone arrays and corresponding speech enhancement systems can be constructed. The first microphone array consists of one omni-microphone collocated with one fiber acoustic sensor. And the second one consists of two collocated fiber acoustic sensors with orthogonal dipole beam patterns. By using the first microphone array, a first-order adaptive beamformer is designed in this thesis to reduce speech interference effects and separate speeches. In this design, an adaptive first-order beam pattern is formed by means of normalized least mean square method. Considering a scenario where the desired speech and interference speech are present at the same time, this adaptive beamformer is able to point the null angle of beam pattern at the undesired speaker to achieve speech interference reduction. In order to verify this idea, numerical simulations are conducted in an ideal condition (clean speech without reverberation) and real scenario (clean speech corrupted by white noise and reverberation). The results show that this design is able to improve speech quality significantly in ideal case. Under the condition suffering from white noise and reverberation, the improvement is achieved as well but at a much smaller scale. By using the second collocated microphone array, a speech enhancement system is proposed to make the collocated fiber acoustic sensors be able to capture speech from any directions. This system includes three main parts. The first part conducts DOA (direction of arrival) estimation empowered by a machine learning method. Here the inter-channel acoustic intensity difference is employed to compute raw DOA estimates with the presence of white noise and reverberation. After obtaining the raw DOA estimates, the machine learning method (wrapped Gaussian mixture model) is used to give a more accurate DOA estimation. This proposed method is robust to both white noise and reverberation with a low computational complexity and solves the phase ambiguity problem (0 and π are identical). In the second part, by using the orthogonality of the dipoles of the two collocated fiber acoustic sensors (one is sin⁡θ and the other is cos⁡θ), along with the DOA (θ) estimated by the wrapped Gaussian mixture model, a steerable dipole beam pattern is generated to point the main lobe at the speaker. In the third part, a noise reduction procedure is applied to the output signal of the steerable beamformer. The proposed method is based on a time-frequency mask, which is used to filter out time-frequency bins of white noise and keep those of speech signal. In order to verify the effectiveness of the designed system, numerical simulations are conducted in the existence of both white noise and reverberation. The result shows that the proposed DOA estimation method is robust to both white noise and reverberation. It implies that this type of microphone array is able to obtain precise speaker spatial information. Meanwhile, the audio quality of the output signal of this system is improved by at least 50%

Distributed Beamforming of Two Autonomous Transmitters

The distributed beamformer is a scheme which provides spatial diversity to combat the undesired effects of the wireless channel. The distributed beamformer requires strict carrier frequency and phase synchronization in order to maximize SNR at a destination for fixed transmit powers. This project investigated the synchronization of two such transmitters in a wired single path channel with off-the-shelf integrated circuits. Additionally, a stable hardware platform for an acoustic (wireless) implementation of such a distributed beamformer was provided

USE OF MICROPHONE DIRECTIVITY FOR THE LOCALLIZATION OF SOUND SOURCES

In a recent paper [1] the proof-of-concept of a novel approach for the localization of sound source was demonstrated. The method relies on the use of unidirectional microphones and amplitude-based signals' features to extract information about the direction of the incoming sound. By intersecting the directions identified by a pair of unidirectional microphones, the position of the emitting source can be identified.In this study we expand the work presented in that paper by assessing the effectiveness of the approach for the localization of an acoustic source in an indoor setting. As the method relies on the accurate knowledge of the microphones directivity, analytical expression of the acoustic sensors polar pattern were derived by testing them in an anechoic chamber. Then an experiment was conducted in a classroom-type environment by using an array of three unidirectional microphones. The ability to locate the position of a commercial speaker placed at different position is discussed.It is believed that this method may pave the road toward a new generation of reduced size sound detectors and localizers

Duct acoustics for air-coupled ultrasonic phased arrays

Air-coupled ultrasound is used in many applications such as range inding, tactile feedback, flow metering or non-destructive testing. The transducers directivity is a crucial acoustic property for all these applications. For instance, a narrow beam width allows for higher angular resolutions, whereas a wider beam width allows for emitting the sound wave in a bigger area for obstacle detection. However, the transducers dimensions influence its directivity and its resonance frequency. In order to decouple the acoustic aperture from transducer, acoustic waveguides are investigated in this work. This way, grating lobe free phased arrays can be built for unambiguous beamforming. In this thesis, the wave propagation inside these waveguides, including coupling mechanisms from the transducer till the free-field, are investigated. First, the state of the art of duct acoustics applications in the audible range and the ultrasound range are presented. Afterwards, different duct acoustics models are derived and compared. Each model is validated for 40 kHz, a duct length of 80 mm and an aperture between 10 mm and 3.4 mm. The challenge of the simulations is to take higher modes into account while reducing the calculation times. Therefore, analytical and numerical models were investigated. As a result, the boundary element method is the most efficient approach for the given geometry wavelength ratio using the commercial software COMSOL Multiphysics. With this method, free-field calculations on a single Xeon E5-2660 v3 CPU and 256 GB RAM without the need of a cluster are possible. The model is validated with calibrated measurements in an anechoic chamber. Therefore, an automated measurement system is established where a calibrated measurement microphone moves relative to the transducer, thus characterizing the sound field in front of the transducer. This setup can measure a hemisphere with a radius of up to 6 m and has a dynamic range of 111 dB. After the validation of the numerical model, waveguide geometry optimizations were conducted. The analyzed properties were: the influence of a perpendicular output and input surface on the wave propagation inside the waveguide; the size of the output aperture; length variations of the waveguide including temperature dependence; the position of tapering and types of losses due to the waveguide. As a result, the perpendicular input is crucial for fundamental mode propagation, otherwise higher modes occur, because the input diameter is bigger compared to the wavelength. The size of the output surface can be increased for line arrays with an SPL gain of +10 dB. However, the limit of the aperture size is 3.7 × λ, otherwise higher modes occur at the output which lead to defocusing of the main lobe. The length of the waveguide can increase the SPL. However, the industrial temperature demands of −25◦C to 75◦C have the same influence on the SPL as the length optimization (±4.8 dB), and, thus, are not investigated in more detail. The positioning of the tapering has just a minor influence of ±0.4 dB. The losses of the waveguide are −10 dB with diffraction loss as the dominant part. The losses inside the waveguide (reflection and thermoviscous losses) could not be validated with measurements due to the narrow bandwidth of the transducers, since the incident and reflected wave superposed. The derived results of the geometry optimization were used to build four line arrays. First, a waveguide with equal length ducts was built as a reference. Second, a Bézier waveguide with plane input surfaces for the transducers was designed. Third, the output aperture was changed from round outputs to rectangular shapes to increase the SPL and sensitivity. Last, a shortened version of the Bézier waveguide was built which has a reduced length of 65%. All four waveguides were simulated using the boundary element method and validated with the measurement etup. As a result, in both simulation and measurement the shorten waveguide has an increased SPL of +5 dB compared to the reference waveguide with equal length ducts. Thus, it is possible to build compact waveguides for air-coupled phased arrays. Next, the influence of different duct lengths in an acoustic waveguide is analyzed in more detail. Using ducts of different length offers more design freedom for the entire waveguide for compact design, easier assembly and reduced assembly time. However, different lengths must be compensated with additional time delays. Therefore, two waveguides were compared. First, an equal length waveguide was used. Second, a waveguide with Bézier-shaped ducts was used. The time delays, due to varying duct lengths, were measured and simulated with analytic and numerical methods. Afterwards, the directivity patterns of both waveguides were compared. As a result, the time compensation has no significant impact on the beam profile regarding side lobe level and half power beam width. In addition, SPL deviation of the waveguides are within the manufacturing tolerances of the transducers. The last aspect investigated in this thesis is the water resistance of the waveguide. Since it is designed for air-coupled ultrasound, it can be clogged due to dirt, dust or liquid. Two commonly known solutions for this issue is the use of hydrophobic fabrics or thin films. Therefore, both solutions were compared. First, these two approaches showed no significant impact on the beamforming capabilities of the phased array. In addition, the IP class of the fabric reached IPX7 and the thin film achieved even IPX8. Furthermore, the fabric has a minor insertion loss of just −1.8 dB. In contrast, the film reduces the SPL by −7.5 dB. This loss can be further reduced with special effort to +0.4 dB by changing the waveguide geometry and tuning the system to the correct resonance frequency. However, this shows that the film has a high temperature dependence compared to the fabric. In conclusion, acoustic waveguides enhance the acoustic properties of ultrasonic sensors. The directivity can be decoupled from the transducer and customized for a certain application

Vector sensors for underwater : acoustic communications

Acoustic vector sensors measure acoustic pressure and directional components separately. A claimed advantage of vector sensors over pressure-only arrays is the directional information in a collocated device, making it an attractive option for size-restricted applications. The employment of vector sensors as a receiver for underwater communications is relatively new, where the inherent directionality, usually related to particle velocity, is used for signal-to-noise gain and intersymbol interference mitigation. The fundamental question is how to use vector sensor directional components to bene t communications, which this work seeks to answer and to which it contributes by performing: analysis of acoustic pressure and particle velocity components; comparison of vector sensor receiver structures exploring beamforming and diversity; quanti cation of adapted receiver structures in distinct acoustic scenarios and using di erent types of vector sensors. Analytic expressions are shown for pressure and particle velocity channels, revealing extreme cases of correlation between vector sensors' components. Based on the correlation hypothesis, receiver structures are tested with simulated and experimental data. In a rst approach, called vector sensor passive time-reversal, we take advantage of the channel diversity provided by the inherent directivity of vector sensors' components. In a second approach named vector sensor beam steering, pressure and particle velocity components are combined, resulting in a steered beam for a speci c direction. At last, a joint beam steering and passive time-reversal is proposed, adapted for vector sensors. Tested with two distinct experimental datasets, where vector sensors are either positioned on the bottom or tied to a vessel, a broad performance comparison shows the potential of each receiver structure. Analysis of results suggests that the beam steering structure is preferable for shorter source-receiver ranges, whereas the passive time-reversal is preferable for longer ranges. Results show that the joint beam steering and passive time-reversal is the best option to reduce communication error with robustness along the range.Sensores vetoriais acústicos (em inglês, acoustic vector sensors) são dispositivos que medem, alem da pressão acústica, a velocidade de partícula. Esta ultima, é uma medida que se refere a um eixo, portando, esta associada a uma direção. Ao combinar pressão acústica com componentes de velocidade de partícula pode-se estimar a direção de uma fonte sonora utilizando apenas um sensor vetorial. Na realidade, \um" sensor vetorial é composto de um sensor de pressão (hidrofone) e um ou mais sensores que medem componentes da velocidade de partícula. Como podemos notar, o aspecto inovador está na medição da velocidade de partícula, dado que os hidrofones já são conhecidos.(...)This PhD thesis was supported by the Brazilian Navy Postgraduate Study Abroad Program Port. 227/MB-14/08/2019

Advanced Radio Frequency Antennas for Modern Communication and Medical Systems

The main objective of this book is to present novel radio frequency (RF) antennas for 5G, IOT, and medical applications. The book is divided into four sections that present the main topics of radio frequency antennas. The rapid growth in development of cellular wireless communication systems over the last twenty years has resulted in most of world population owning smartphones, smart watches, I-pads, and other RF communication devices. Efficient compact wideband antennas are crucial in RF communication devices. This book presents information on planar antennas, cavity antennas, Vivaldi antennas, phased arrays, MIMO antennas, beamforming phased array reconfigurable Pabry-Perot cavity antennas, and time modulated linear array