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The blind source separation problem has received a great deal of attention in previous years. The aim of this problem is to estimate a set of original source signals from a set of linearly mixed signals through any number of signal processing techniques. While many methods exist that attempt to solve the blind source separation problem, a new technique is being used that uniquely separates audio sources as they are received from a microphone array. In this thesis a new algorithm is proposed that that utilizes the ICA algorithm in conjunction with a filtering technique that separates source signals and then removes sources of interference so that a signal of interest can be accurately tracked. Experimental results will compare a common blind source separation technique to the new algorithm and show that the new algorithm can detect a signal of interest and accurately track it as it moves through an anechoic environment. / Master of Science

This thesis describes the design and implementation of a 4-channel microphone array that is an adaptive beamformer used for hands-free telephony in a noisy environment. The microphone signals are amplified, then sent to an A/D converter. The microprocessor board takes the data from the 4 channels and utilizes digital signal processing to determine the direction-of-arrival of the sources and create an output which 'steers' the microphone array to the desired look direction while trying to minimize the energy of interference sources and noise. All of the processing for this thesis will be done on a computer using MATLAB. The MUSIC algorithm is used for direction finding in this thesis. It is shown to be effective in estimating direction-of-arrival for 1 speech source and 2 speech sources that are spaced fairly apart, with significant results down to a -5 dB SNR even. The MUSIC algorithm requires knowledge of the number of sources a priori, requiring an estimator for the number of sources. Though proposed estimators for the number of sources were examined, an effective estimator was not encountered for the case where there are multiple speech sources. Beamforming methods are examined which utilize knowledge of the source direction-of-arrival from the MUSIC algorithm. The input is split into 6 subbands such that phase-steered beamforming would be possible. Two methods of phase-steered beamforming are compared in both narrowband and wideband scenarios, and it is shown that phase-steering the array to the desired source direction-of-arrival has about 0.3 dB better beamforming performance than the simple time-delay steered beamformer using no subbands. As the beamforming solution is inadequate to achieve desired results, a generalized sidelobe canceler (GSC) is developed which incorporates a beamformer. The sidelobe canceler is evaluated using both NLMS and RLS adaptation. The RLS algorithm inherently gives better results than the NLMS algorithm, though the computational complexity renders the solution impractical for implementation with today's technology. A testing setup is presented which involves a linear 4-microphone array connected to a DSP chip that collects the data. Tests were done using 1 speech source and a model of the car noise environment. The sidelobe canceler's performance using 6 subbands (phase-delay GSC) and using 1 band (time-delay GSC) with NLMS updating are compared. The overall SNR improvement is determined from the signal and noise input and output powers, with signal-only as the input and noise-only as the input to the GSC. The phase-delay GSC gives on average 7.4 dB SNR improvement while the time-delay GSC gives on average 6.2 dB SNR improvement. / Master of Science

The complex relationship between array gain patterns and microphone distributions limits the application of traditional optimization algorithms on irregular arrays, which show enhanced beamforming performance for human speech capture in immersive environments. This work analyzes the relationship between irregular microphone geometries and spatial filtering performance with statistical methods. Novel geometry descriptors are developed to capture the properties of irregular microphone distributions showing their impact on array performance. General guidelines and optimization methods for regular and irregular array design are proposed in immersive (near-field) environments to obtain superior beamforming ability for speech applications. Optimization times are greatly reduced through the objective function rules using performance-based geometric descriptions of microphone distributions that circumvent direct array gain computations over the space of interest. In addition, probabilistic descriptions of acoustic scenes are introduced to incorporate various levels of prior knowledge for the source distribution. To verify the effectiveness of the proposed optimization methods, simulated gain patterns and real SNR results of the optimized arrays are compared to corresponding traditional regular arrays and arrays obtained from direct exhaustive searching methods. Results show large SNR enhancements for the optimized arrays over arbitrary randomly generated arrays and regular arrays, especially at low microphone densities. The rapid convergence and acceptable processing times observed during the experiments establish the feasibility of proposed optimization methods for array geometry design in immersive environments where rapid deployment is required with limited knowledge of the acoustic scene, such as in mobile platforms and audio surveillance applications.

<div>Fluid power is widely used in a variety of applications such as construction machines, aerospace, automotive, agricultural machinery, manufacturing, etc. Although this technology has many obvious advantages such as compactness, robustness, high power density, and so forth, there is much room for improvement, of which one of the most important and challenging problems is the noise.</div><div><br></div><div>Different institutes have been researching fluid power noise for decades. However, much of the experimental investigation was based on simple measurement and analysis techniques, which left the designers/researchers no method of understanding the complicated phenomena. A microphone array is a powerful tool that unfortunately has not been introduced to the fluid power noise research. By capturing the magnitude and phase information in space, a microphone array enables the noise source identification, separation, localization and so forth.</div><div><br></div><div>This thesis focuses on implementing the microphone array processing techniques on a synthetic microphone array for fluid power noise diagnostics. Differing from traditional scan-based approaches, the synthetic array is created by synchronizing the non-synchronous measurements to achieve the equivalent effect of a multi-microphone snapshot. The final results will show the power of microphone arrays and provide an economical solution to achieve approximate results when a real microphone array is not available.</div>

The geometrical characteristics of acoustical radiation are of great importance in instrument design and synthesis, and multiple simplified models have been developed in the past to describe them. In this work two experimental methodologies are proposed and carried out, studying the frequency-dependent radiation in a collection of popular brass instruments with different grades of flaring, and making use of the axis-symmetry of these instruments. The first method uses a scanning linear array and is carefully designed to extract the linear properties of the radiation field. The results of this experimental method are a database of impulse responses distributed in space, and effectively covering a bidimensional on-axis section of the radiation field approximately 0.6 m by 0.9 m. These data can then be used for the validation of a number of simplified physical models used to describe the radiation of these types of instruments. The second method aims at visualising radiation for high amplitude excitation, where shock waves are generated inside the instrument due to non-linear propagation of the plane wave. In this case, the experimental methodology used, taking advantage of the strong density and temperature gradients generated in the air, is an on-axis schlieren optical system. General results of this visualisation show a strong increase in focused directivity at high frequencies and loud playing dynamics, due to the spectral enrichment typical of this family of instruments. The second section of this thesis focuses on the study of playing gestures in the trombone, and could also be applicable to other slide instruments. During glissando playing in the trombone the length of the cylindrical slide section within the bore is altered while waves are propagating. Slide velocities of 2 metres per second are not unusual and result in a (small but measurable) Doppler shift in the wave coming from the mouthpiece before it arrives at the bell. An additional effect is observed in terms of the volume of air within the instrument changing telescopically, leading to a localised change in DC pressure and a resulting flow, which generates infrasound components within the bore. The effects of these playing gestures are investigated in two different setups; one with a high frequency sinusoidal excitation generated by a compression driver, and another one using an artificial mouth to play the instrument. In both experiments the pressures at the mouth or mouthpiece, water key and bell were tracked using microphones and the position of the slide was tracked using a laser distance sensor. Both Doppler shifting and infrasound components were detected for both experimental setups, although the effect on a soft termination such as the artificial lips requires further examination.

Bioacoustics is a relatively new field of research focused on studying the acoustic signals of vocal animal species. The field has been a topic of interest for many years due its passive approach and avoidance of species-level limitations, such as tracking rare or nocturnal species. It has been used to locate birds in terrestrial environments; however, localization in urban environments remains unstudied. This research aims to fill the gap by attempting to estimate the location of 30 discrete calls in eight unique, urban environments. Sites represented two distinct traffic scenarios: moderate traffic and high traffic. Three system arrays of three different sizes utilizing the Song Meter SM2+ units were tested at each site to determine the effect of array size on call visibility and location estimation. An American robin (Turdus migratorius) distress call was played through a loudspeaker at the thirty locations for each array. The spectrogram of each of these calls was examined to determine the number of channels with a visible call signature. If the file contained at least one visible call per song meter (36% of our sound files), cross correlation was used to determine the differences in the time of arrival of calls at all the microphones in the array, called lag values, which were used to calculate the origin location of the call. However, resulting lag values in this study were too large to produce reliable location estimates. This was likely due to imprecise synchronization in the field or poorly defined calls within the spectrograms. Our overall low visibility is likely a result of the high signal to noise ratio common in urban environments. Further research is necessary to continue to test the viability of acoustic localization in urban environments.

No description available.

The detection of sound sources with microphone arrays can be enhanced through processing individual microphone signals prior to the delay and sum operation. One method in particular, the Phase Transform (PHAT) has demonstrated improvement in sound source location images, especially in reverberant and noisy environments. Recent work proposed a modification to the PHAT transform that allows varying degrees of spectral whitening through a single parameter, andamp;acirc;, which has shown positive improvement in target detection in simulation results. This work focuses on experimental evaluation of the modified SRP-PHAT algorithm. Performance results are computed from actual experimental setup of an 8-element perimeter array with a receiver operating characteristic (ROC) analysis for detecting sound sources. The results verified simulation results of PHAT- andamp;acirc; in improving target detection probabilities. The ROC analysis demonstrated the relationships between various target types (narrowband and broadband), room reverberation levels (high and low) and noise levels (different SNR) with respect to optimal andamp;acirc;. Results from experiment strongly agree with those of simulations on the effect of PHAT in significantly improving detection performance for narrowband and broadband signals especially at low SNR and in the presence of high levels of reverberation.

A method to synthesize the sound radiation characteristics of musical instruments in a wave field synthesis (WFS) system is proposed and tested. Radiation patterns of a violin are measured with a circular microphone array which consists of 128 pressure receivers. For each critical frequency band one exemplary radiation pattern is decomposed to circular harmonics of order 0 to 64. So the radiation characteristic of the violin is represented by 25 complex radiation patterns. On the reproduction side, these circular harmonics are approximated by 128 densely spaced monopoles by means of 128 broadband impulses. An anechoic violin recording is convolved with these impulses, yielding 128 filtered versions of the recording. These are then synthesized as 128 monopole sources in a WFS system and compared to a virtual monopole playing the unfiltered recording. The study participants perceive the tone color of the recreated virtual violin as being dependent on the listening position and report that the two source types have a different ‘presence’. The test persons rate the virtual violin as less natural, sometimes remarking that the filtering is audible at high frequencies. Further studies with a denser spacing of the virtual monopoles and a presentation in an anechoic room are planned.