Although it is widely accepted that the diffusion of early reflections in acoustic spaces intended for music performance greatly improves the perceived quality of sound, current manufacturers of synthetic reverberation engines continue to model reflecting surfaces as having almost perfectly specular characteristics. This dissertation describes a hybrid method of simulating diffusion based on both physical and phenomenological modeling components. / In 1979, Manfred Schroeder described a method of designing and constructing diffusing surfaces based on a rather simple mathematical algorithm which provides diffused reflections in predictable frequency bands. This structural device, now known as a "Schroeder diffuser," has become a standard geometry used in constructing diffusive surfaces for spaces intended for music rehearsal, recording and performance. While it is possible to use DSP to model the characteristics of reflections off such a surface, a reflection model based exclusively on a surface constructed of a Schroeder diffuser has proven in informal tests to be as aesthetically inadequate as a perfectly specular model. Control of both the spatial and temporal envelopes of the diffusive reflection are required by an end user in order to tailor the reflection characteristics to the desired impression. / In 1974 an empirical model for computing light reflections off objects in a three-dimensional environment was developed by Phong Bui-Toung. This algorithm incorporated both a specular and diffuse component with relationships controlled by an end user. / This dissertation describes the adaptation and implementation of the Phong shading algorithm in conjunction with a physical model of components of the Schroeder diffuser for the modeling of diffuse reflections in synthetic acoustic environments. The inclusion of the Phong algorithm provides precise control over the balance between the spectral and diffusive components of the reflection. In addition, directivity functions for sound sources and receivers in the virtual space are described. / Analysis and evaluation of the model using mathematical and empirical methodologies are discussed and stereo and multichannel audio examples produced by the system are included.

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This thesis details the performance of the reverberation chamber of the Integrated Acoustics Laboratory (IAL), equipped with experimental lightweight diffusers. Reverberation chambers are generally equipped with dense baffles, called diffusers, which are designed to reflect but not absorb sound, in an effort to create a sound field in the chamber with uniform energy density. Industry standards, such as ASTM C423, ISO 354, and ISO 3741 for sound absorption and sound power testing in reverberation chambers, recommend the use of stationary and rotating diffusers, made of a material with high surface density and low absorption. Instead, lightweight fiberglass diffuser panels were installed in the IAL reverberation chamber because they are safer, less expensive and more flexible; their performance in the IAL chamber was evaluated. Preliminary testing of the IAL instrumentation chain and analysis techniques documented their acceptable performance. Qualification testing per the abovementioned standards proved that the IAL chamber, equipped with stationary lightweight diffusers, was fit for testing sound power but not sound absorption. However, when equipped with a combination of stationary and rotating lightweight diffusers, the chamber qualified for sound absorption tests. Optimization of absorption testing methodology showed that specimen area did not significantly affect the measured sound absorption coefficient unless the specimen was highly absorptive or the area was significantly less than the recommended 6.69 m2. Also, increasing the empty room absorption of the acoustically hard IAL chamber did not improve the reproducibility of absorption measurements. With regard to length of test, absorption tests in the IAL chamber should include the measurement of 225 decays to attain the representative repeatability values of ASTM C423 for frequencies 315 Hz and higher. Comparative absorption testing showed that the chamber reproduced sound absorption results well; when round robin testing was replicated in the chamber, results were not statistically different from other laboratories. However, the reproducibility was worse for highly absorptive specimens. Sound power testing produced highly reproducible results, well within the limits of reproducibility of the standard. It can be concluded that a combination of stationary and rotating lightweight diffusers made the IAL chamber fit for sound absorption and sound power testing.

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Bibliography: leaves 140-149 / x, 148 leaves : ill ; 30 cm. / Title page, contents and abstract only. The complete thesis in print form is available from the University Library. / Thesis (Ph.D.)--University of Adelaide, Dept. of Mechanical Engineering, 1989

Thesis (Elec. E.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and, (M.S.)--Massachusetts Institute of Technology, Dept. of Ocean Engineering, 1976. / Microfiche copy available in Archives and Engineering. / Includes bibliographical references. / by James B. Gallemore. / M.S.

M.Ing. (Electrical and Electronic Engineering) / Acoustic reverberation is put under the spotlight. A review of the theory was conducted followed by a look into digital artificial reverberation. Measurement methodology is presented including a review of the recently published ISO standard pertaining to reverberation. Experimental testing was conducted for four acoustically different environments with one of them almost completely anechoic. The reverberation characteristic of these four environments were measured and analysed according to the relevant ISO standards. The results were then used in a further study of digital artificial reverberation applied to impulse and vocal sounds. The anechoic sounds were artificially reverberated using Cool Edit Pro software to mimic the sound obtained that had natural reverberation present. The focus was on the RT as well as the EDT of the decay slope. The artificial method of applying reverberation was evaluated using two methodologies, firstly objective methods relying on mathematics; secondly, by subjective personal evaluations using a statistical analysis of a listening test questionnaire. Both the objective and subjective results confirmed that digital artificial reverberation methods could be applied successfully to impulse sounds and vocals. The results provide a basis for the motivation of computerised methods in the studio recording process especially for rooms that are moderately anechoic.

Active control of sound transmission through an elastic plate placed between two reverberation chambers is studied experimentally. Active acoustic control is performed using piezoelectric sensors and actuators bonded to the plate. The control technique uses an adaptive control algorithm. Results are presented for harmonic excitation provided by a speaker in the source chamber at two resonant frequencies of the plate. Influence of different types of error sensors, varied actuator locations, and varied speaker locations are studied. Compared to microphone sensors in the receiving chamber, piezoelectric sensors are shown to be effective in reducing sound transmission through the plate. Average reduction of sound pressure level (SPL) on the order of 20 dB or 13 dB are achieved when the plate vibrates at mode (3,1) or (3,3). Microphone sensor locations are shown to influence the controlled sound field, those located where the direct sound field is dominant result in larger SPL reductions. SPL reductions are caused by two mechanisms: modal reduction and modal restructuring, and the dominance of either is shown to depend on actuator locations. When the sound field is non-diffuse, speaker locations influence the SPL and the SPL reduction by changing the plate's structural response. Also included in this work, previously developed one-dimensional (I-D) modal sensor theory for beams is used to develop modal sensors for a clamped plate. Two I-D modal sensors are applied to a fully clamped plate and each shown to observe a particular subset of plate vibration modes. Previous work developed the theory for two-dimensional (2-D) modal sensors for simply-supported plates. A necessary and sufficient condition for the spatial functions of 2-D modal sensors are developed for plates with arbitrary boundary conditions. / Master of Science

In this thesis, we tackle key challenges in processing audio signals, specifically focusing on speech and music. These signals are crucial for human interaction with both the environment and machines. Our research addresses three core topics: speech denoising, speech dereverberation, and music-dance generation, each of which plays a vital role in enhancing the harmony between audio and human interaction. Leveraging machine learning and human-centric approaches inspired by classical algorithms, we develop methods to mitigate common audio degradations, such as additive noise and multiplicative reverberation, delivering high-quality audio suitable for human use and applications. Furthermore, we introduce a real-time, music-responsive system for generating 3D dance animations, advancing the integration of audio signals with human engagement. The first focus of our thesis is the elimination of additive noise from audio signals by focusing on short pauses, or silent intervals, in human speech. These brief pauses provide key insights into the noise profile, enabling our model to dynamically reduce ambient noise from speech. Tested across diverse datasets, our method outperforms traditional and audiovisual denoising techniques, showcasing its effectiveness and adaptability across different languages and even musical contexts. In the second work of our research, we address reverberation removal from audio signals, a task traditionally reliant on knowing the environment's exact impulse response—a requirement often impractical in real-world settings. Our novel solution combines the strengths of classical and learning-based approaches, tailored for online communication contexts. This human-centric method includes a one-time personalization step, adapting to specific environments and human speakers. The two-stage model, integrating feature-based Wiener deconvolution and network refinement, has shown through extensive experiments to outperform current methods, both in effectiveness and user preference. Transitioning from foundational audio signal enhancement and analysis to a more dynamic realm, our research culminates in a novel, interactive system for real-time 3D human dance generation. Contrasting with the passive human-centric assumptions of our previous works, this final work actively engages users, enabling direct interaction with a system that synchronizes expressive dance movements to live music, spanning various musical elements like type, tempo, and energy. This innovative approach, diverging from traditional choreography methods, leverages spontaneous improvisation to generate unique dance sequences. These sequences, a mix of pre-recorded choreographies and algorithm-generated transitions, adapt to real-time audio inputs, offering customization through personal 3D avatars. This system's user-centric design and interactivity are validated by user studies, confirming its effectiveness in creating an immersive and engaging user experience.