This thesis project aims for the development of a demonstration platform on mobile devices for testing and demonstrating algorithms for 2D and 3D spatial sound reproduction. The demo system consists of four omnidirectional microphones in a square planar array, an Octo sound card (from Audio Injector), a Raspberry Pi 3B+ (R-Pi) single-board computer and an inertial measurement unit (IMU) located in the center of the array. The microphone array captures sound, which is then digitized, and in turn, transferred to the R-Pi. On the R-Pi, the digitized sound signal is rendered through the directional audio coding (DirAC) algorithm to maintain the spatial properties of the sound. Finally, the digital signal and spatial properties are rendered through Dirac VR to maintain a spatial stereo signal of the recorded environment. The directional audio coding algorithm was initially implemented in Matlab and then ported to C++ since the R-Pi does not support Matlab natively. The ported algorithm was verified on a four-channel in and six-channel out system, processing 400 000 samples at 44 100 kHz. The results show that the C++ DirAC implementation maintained a maximum error of 4.43e-05 or -87 dB compares to the original Matlab implementation. For future research on spatial audio reproduction, a four-microphone smartphone mock-up was constructed based on the same hardware used in the demo system. A software interface was also implemented for transferring the microphone recordings and the orientation of the mock-up to Matlab.

To date electric pianos and samplers tend to concentrate on authenticity in terms of temporal and spectral aspects of sound. However, they barely recreate the original sound radiation characteristics, which contribute to the perception of width and depth, vividness and voice separation, especially for instrumentalists, who are located near the instrument. To achieve this, a number of sound field measurement and synthesis techniques need to be applied and adequately combined. In this paper we present the theoretic foundation to combine so far isolated and fragmented sound field analysis and synthesis methods to realize a radiation keyboard, an electric harpsichord that approximates the sound of a real harpsichord precisely in time, frequency, and space domain. Potential applications for such a radiation keyboard are conservation of historic musical instruments, music performance, and psychoacoustic measurements for instrument and synthesizer building and for studies of music perception, cognition, and embodiment.

No description available.

京都大学 / 0048 / 新制・課程博士 / 博士(情報学) / 甲第18412号 / 情博第527号 / 新制||情||93(附属図書館) / 31270 / 京都大学大学院情報学研究科知能情報学専攻 / (主査)教授 奥乃 博, 教授 河原 達也, 准教授 CUTURI CAMETO Marco, 講師 吉井 和佳 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM

No description available.

Phased microphone arrays are valuable tools for aeroacoustic measurements that can measure the directivity of multiple acoustic sources. However, when deployed in closed test-section wind tunnels, the acoustics suffer due to intense pressure fluctuations contained in the wall-bound turbulent boundary layer. Furthermore, phased microphone arrays require many sensors distributed over a large aperture to ensure good spatial resolution over a wide frequency range. Microphone arrays of such large count are not always feasible due to constraints in space and cost. This thesis describes an alternative approach for measuring single broadband acoustic sources that uses an acoustic metasurface. The metasurface is comprised of a meandering channel of quarter-wave cavities and an array of equally spaced half-wave open through-cavities. A series of tests were conducted in Virginia Tech's Anechoic Wall-Jet Tunnel where combinations of a wall-bound turbulent jet-flow and a single broadband acoustic source were used to excite the metasurface and produce acoustic surface waves. Measurements of the acoustic surface waves were performed using two methods: a pair of traversing microphones scanning the pressure field along the length of the metasurface 0.25 mm beneath its bottom face, and an array of unequally spaced microphones embedded inside the metasurface. Spectral analysis on the measurements revealed that the inclusion of multiple through-cavities leads to constructive reinforcement of select acoustic surface waves as a function of the acoustic source location. In the case of the embedded microphones, acoustic beamforming was applied in order to extract spatial information. This reinforcement was observed during measurements made with both flow and acoustic excitation, up to Wall-Jet Tunnel nozzle exit speeds of 40 m/s beyond which it was no longer seen. A series of quiescent measurements made with a range of speaker locations constituted a calibration for the metasurface which was used to locate an unknown broadband acoustic source within an The Root-Mean-Square (RMS) error of 1.06 degrees. / Master of Science / Phased microphone arrays are valuable tools for aeroacoustic measurements that can measure the directivity of multiple acoustic sources within a sound field. When used in conjunction with signal processing techniques, such as delay-and-sum beamforming, a researcher or engineer can obtain an intuitive view of the sound field and distinguish between multiple sources over a wide frequency range. However, these microphone arrays often utilize dozens of microphones which raises the array's complexity and cost. Furthermore, when a phased microphone array is mounted flush to the wall of a wind tunnel test section, it is submerged under a turbulent boundary layer which imposes intense pressure fluctuations on the microphones making it difficult to identify acoustic sources. Boundary layers form at the interface between a fluid and solid interface. This thesis describes experimentation performed in the Virginia Tech Anechoic Wall-Jet Tunnel on a new type of pressure sensing microphone array that leverage acoustic metamaterial technology. The acoustic metamaterial shields the microphones from the flow, lessening the influence of the turbulent boundary layer on the measurement. The focus in this thesis is on the novel array's ability to locate a single broadband acoustic source using as few as six microphones. The metasurface was installed in the Wall-Jet Tunnel test plate such that an array of evenly spaced through-cavities are flush to the surface. The through-cavities communicate the pressure field on top of the test surface to a meandering channel of interconnected closed cavities below. Near the resonant depth frequencies of the closed cavities, acoustic surface waves form which are evanescent pressure waves that are bound to the surface or structure that support them. The interference between the acoustic surface waves generated at each through-cavity leads to reinforced acoustic surface waves which are sensitive to the direction of a broadband source. In all, an acoustic metamaterial was tested under a variety of conditions such as: Wall-Jet Tunnel flow speed, speaker location, and the number of through-cavities open. The performance of the novel array and future plans are discussed.

京都大学 / 新制・課程博士 / 博士(情報学) / 甲第23309号 / 情博第745号 / 新制||情||127(附属図書館) / 京都大学大学院情報学研究科知能情報学専攻 / (主査)准教授 吉井 和佳, 教授 河原 達也, 教授 西野 恒, 教授 田中 利幸 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM

No description available.

Two rotor configurations with different non-uniform inflows were studied: a rotor ingesting the wake of an upstream cylinder and a rotor ingesting a thick axially symmetric boundary layer from an upstream centerbody. In both cases, the undisturbed inflow was measured without the rotor present in order to characterize the inflow, in particular to calculate the unsteady upwash velocity distribution at the location of the rotor. In addition, detailed acoustic measurements were completed using a 251-channel large-area microphone array. In all, over 400 conditions covering different advance ratios, angles of yaw, and inflow conditions were measured. Measurements of the sound show that the source has a complex directivity, different from that of a streamwise aligned dipole, due to the inhomogeneous unsteady upwash distribution. In addition, observers at different far field locations will perceive sources from different locations on the rotor disk. The directivity is a function of both the rotor geometry and turbulent inflow. A simplified model of the sound source was developed using these inputs and accurately predicts trends observed in the far field noise. For the cylinder wake ingestion case, on-blade measurements of the flow field show that the wake is drawn to the center of the rotor disk with increasing thrust. This is particularly noticeable if the wake does not strike the center of the rotor disk. The effects of this flow distortion on the far field directivity are well predicted by the model. The effects of yaw to rotate the produced sound field can be inferred from this model as well. A novel beamforming procedure was used to isolate sources across the face of the rotor for the cylinder wake ingestion case for an upstream observer position. This method may be used to isolate different sound sources on a rotor if multiple sources are present or if different regions of the rotor disk need to be isolated. The directivity of a rotor ingesting an axially symmetric boundary layer is far less complex than the ingestion of a two-dimensional cylinder wake, but measurements still show the perceived source location shift with observer location. Overall, the proposed noise modeling technique is an efficient method to predict the directivity of turbulence ingestion noise for inhomogeneous inflows. This can enable quick absolute noise predictions at all far field locations using only a single point measurement or far field noise prediction to establish absolute levels. / Doctor of Philosophy / In many engineering applications, rotors interact with turbulence. Aircraft and ships with rear mounted propellers can have upstream appendages or discontinuities that generate turbulence that travels downstream and is drawn into the propeller. Wind turbines interact with turbulence in the atmosphere and with turbulent wakes from other turbines. Interaction of a rotor with turbulence results in unsteady loading on the rotor blades that can radiate as sound, causing unwanted community noise or vehicle detection. As such, prediction and reduction of noise due to turbulence ingestion is highly desirable and remains an active area of research. Turbulence ingestion noise is well understood from first principles and can be successfully predicted provided an accurate description of the turbulent inflow and unsteady aerodynamic response of the rotor blades. Much work has focused on homogenous, isotropic turbulence ingestion noise, however, in practical applications, the rotor inflow is often non-uniform, anisotropic, and can change dramatically with the thrusting condition of the rotor. Research efforts to develop noise predictions considering these more complex, but practical inflows have focused on the inflow modeling and measurement and have relied on a small subset of sound measurements for validation. The present study seeks to provide new physical insight into inhomogeneous, anisotropic turbulence ingestion noise through wind tunnel experiments. In particular, two rotor configurations with different practical non-uniform inflows are studied: a rotor ingesting the wake of an upstream cylinder and a rotor ingesting a thick axially symmetric boundary layer from an upstream center body. In both cases, the undisturbed inflow was measured without the rotor present in order to characterize the inflow, and detailed acoustic measurements were completed using a 251-channel large-area microphone array. In all, over 400 rotor operating conditions were measured. The acoustic directivity in each case is examined in detail as a function of rotor operating condition. A simplified directivity model is developed and validated with measurements. Ultimately, the directivity model can provide a good engineering approximation of the full directivity with reduced computational time or can be used to extrapolate measured results to positions in the far field where placement of sensors is not possible. The results can also be used to guide the analysis and interpretation of single point or microphone array measurements in the acoustic far field of a rotor.

Time delay estimation (TDE)-based algorithms for estimation of direction of arrival (DOA) have been most popular for use with speech signals. This is due to their simplicity and low computational requirements. Though other algorithms, like the steered response power with phase transform (SRP-PHAT), are available that perform better than TDE based algorithms, the huge computational load required for this algorithm makes it unsuitable for applications that require fast refresh rates using short frames. In addition, the estimation errors that do occur with SRP-PHAT tend to be large. This kind of performance is unsuitable for an application such as video camera steering, which is much less tolerant to large errors than it is to small errors. We propose an improved TDE-based DOA estimation algorithm called time delay selection (TIDES) based on either minimizing the weighted least squares error (MWLSE) or minimizing the time delay separation (MWTDS). In the TIDES algorithm, we consider not only the maximum likelihood (ML) TDEs for each pair of microphones, but also other secondary delays corresponding to smaller peaks in the generalized cross-correlation (GCC). From these multiple candidate delays for each microphone pair, we form all possible combinations of time delay sets. From among these we pick one set based on one of the two criteria mentioned above and perform least squares DOA estimation using the selected set of time delays. The MWLSE criterion selects that set of time delays that minimizes the least squares error. The MWTDS criterion selects that set of time delays that has minimum distance from a statistically averaged set of time delays from previously selected time delays. Both TIDES algorithms are shown to out-perform the ML-TDE algorithm in moderate signal to reverberation ratios. In fact, TIDES-MWTDS gives fewer large errors than even the SRP-PHAT algorithm, which makes it very suitable for video camera steering applications. Under small signal to reverberation ratio environments, TIDES-MWTDS breaks down, but TIDES-MWLSE is still shown to out-perform the algorithm based on ML-TDE. / Master of Science