Synthetic speech is a valuable means of output, in a range of application contexts, for people with visual, cognitive, or other impairments or for situations were other means are not practicable. Noise and reverberation occur in many of these application contexts and are known to have devastating effects on the intelligibility of natural speech, yet very little was known about the effects on synthetic speech based on unit selection or hidden Markov models. In this thesis, we put forward an approach for assessing the intelligibility of synthetic and natural speech in noise, reverberation, or a combination of the two. The approach uses an experimental methodology consisting of Amazon Mechanical Turk, Matrix sentences, and noises that approximate the real-world, evaluated with generalized linear mixed models. The experimental methodologies were assessed against their traditional counterparts and were found to provide a number of additional benefits, whilst maintaining equivalent measures of relative performance. Subsequent experiments were carried out to establish the efficacy of the approach in measuring intelligibility in noise and then reverberation. Finally, the approach was applied to natural speech and the two synthetic speech systems in combinations of noise and reverberation. We have examine and report on the intelligibility of current synthesis systems in real-life noises and reverberation using techniques that bridge the gap between the audiology and speech synthesis communities and using Amazon Mechanical Turk. In the process, we establish Amazon Mechanical Turk and Matrix sentences as valuable tools in the assessment of synthetic speech intelligibility.

Reverberation is the phenomenon when the sound source transmits and causes scattering in active sonar system. This kind of effects often produced in the waveguide, resulting signal interference and signal mask issues. Reverberation can affect the signal to noise ratio, thus understanding the characteristic of environmental reverberation is important. In recent years, there were many studies for the Sizihwan Bay Marine Test Field (MTF), including environmental surveys, Harbor defense and acoustic inversion, however the issue related to reverberation has not been explored. The purpose of this study is to research reverberation in the MTF and focusing on volume reverberation and surface reverberation. In the past, the experiment of Underwater Intruder Detection with active sonar system demonstrated the reverberation in Kaohsiung second harbor. Therefore, this research is about using the experimental data to questions about volume reverberation. Results show, volume reverberation in the port area caused by ship, boundaries, current, impurities in water and biota. On the other hand surface reverberation, this study focusing on numerical simulation, match the results of experimental. Numerical results of RMS height, correlation length and frequency affect the reverberation intensity, but it does not identify the phenomenon of reverberation in experimental results, mainly is the intensity of the source is not enough. This study combined with simulation and experiment, and overviewed the reverberation properties in MTF. Also provided suggestions for following studies.

While humans can separate a sound of interest amidst a cacophony of contending sounds in an echoic environment, machine-based methods lag behind in solving this task. This thesis thus aims at improving performance of audio separation algorithms when they are informed i.e. have access to source location information. These locations are assumed to be known a priori in this work, for example by video processing. Initially, a multi-microphone array based method combined with binary time-frequency masking is proposed. A robust least squares frequency invariant data independent beamformer designed with the location information is utilized to estimate the sources. To further enhance the estimated sources, binary time-frequency masking based post-processing is used but cepstral domain smoothing is required to mitigate musical noise. To tackle the under-determined case and further improve separation performance at higher reverberation times, a two-microphone based method which is inspired by human auditory processing and generates soft time-frequency masks is described. In this approach interaural level difference, interaural phase difference and mixing vectors are probabilistically modeled in the time-frequency domain and the model parameters are learned through the expectation-maximization (EM) algorithm. A direction vector is estimated for each source, using the location information, which is used as the mean parameter of the mixing vector model. Soft time-frequency masks are used to reconstruct the sources. A spatial covariance model is then integrated into the probabilistic model framework that encodes the spatial characteristics of the enclosure and further improves the separation performance in challenging scenarios i.e. when sources are in close proximity and when the level of reverberation is high. Finally, new dereverberation based pre-processing is proposed based on the cascade of three dereverberation stages where each enhances the twomicrophone reverberant mixture. The dereverberation stages are based on amplitude spectral subtraction, where the late reverberation is estimated and suppressed. The combination of such dereverberation based pre-processing and use of soft mask separation yields the best separation performance. All methods are evaluated with real and synthetic mixtures formed for example from speech signals from the TIMIT database and measured room impulse responses.

ABSTRACT For a variety of wireless sensor network applications, sensor nodes may find their received signal strengths dominated by small-scale propagation effects. Particularly impacted are applications designed to monitor structural health and environmental conditions in metal cavities such as aircraft, busses, and shipping containers. Small changes in each sensor’s position or carrier frequency can cause large variations in this received signal strength, thereby compromising link connectivity. We leverage a technique called Wireless Sensors Sensing Wireless (WSSW) in which wireless sensors act as scalar network analyzers in order to characterize their own environment. WSSW data can enable sensors to react to particularly bad fading, such as hyper-Rayleigh, by switching to a good channel or by implementing other mitigation techniques, such as utilizing a diversity antenna. In this work, the WSSW concept has been extended to accommodate mesh networks and include a spectrum analysis capability for recognizing potentially interfering wireless activity. The test of mitigation techniques is often problematic since application sites are far from controlled environments and are often difficult to access. To address this problem, we have developed a Compact Reconfigurable Channel Emulator (CRCE) to create a laboratory environment that is configurable to a variety of repeatable fading scenarios. With the CRCE, fading characteristics found at a specific wireless sensor network location may be replicated inside the chamber to discover the connectivity capabilities of the sensors and the effectiveness of diversity schemes (e.g., channel switching or multi-element antenna arrays).

Os parâmetros acústicos subjetivos são critérios que definem a qualidade acústica de uma sala de música. A apreciação musical dentro da sala é afetada por diversas impressões acústicas que ocorrem ao mesmo tempo. Cada uma dessas impressões é associada a um parâmetro acústico de natureza subjetiva que está correlacionado a uma grandeza física mensurável, constituindo um conjunto de parâmetros acústicos objetivos que formam uma base científica para a análise acústica das salas de música. Neste trabalho desenvolvemos pesquisa de âmbito teórico e experimental envolvendo a análise dos parâmetros acústicos subjetivos mais relevantes para a avaliação da qualidade acústica de salas de escuta musical. Fizemos um levantamento abrangente do material já publicado sobre o assunto, o que nos orientou a respeito das medições acústicas pertinentes à referida análise e nos permitiu formalizar as devidas conclusões. Implementamos e aplicamos a tecnologia necessária para a obtenção dos parâmetros. Determinamos a metodologia experimental mais adequada e efetuamos medições em seis importantes salas de concerto, comparando os resultados. Fizemos uma análise crítica a respeito dos parâmetros acústicos obtidos e aprofundamos a compreensão sobre seus significados e suas utilidades. Por fim, fizemos uma análise subjetiva de júri correlacionando os parâmetros acústicos medidos às respectivas impressões acústicas sobre amostras musicais gravadas nas salas. / Acoustic parameters are the criteria that define the acoustic quality of a music hall. The musical audition inside a room is influenced by a group of acoustic impressions that occur at the same time. Each one of such impressions is associated with a particular subjective parameter that is correlated to a measurable physical value. These values are taken as a set of objective parameters that constitute a scientific basis for the acoustical analysis of a music hall. In this work we have conducted both a theoretical and an experimental investigation on the analysis of the most important acoustic parameters for the evaluation of the quality of a music hall. Also, we have done an extensive research on the related bibliography to support our measurement procedures and formal conclusions. We have implemented the necessary technology to obtain the acoustic parameters. We also have determined the most efficient experimental methodology to carry out acoustic measurements and we have applied this methodology to the measurement of six important concert halls. Then we have compared the data related to each hall and performed a critical analysis of this data, increasing our understanding on the meaning and the usefulness of the acoustic parameters. Finally, we have made a subjective jury analysis, correlating the measured acoustic parameters to the impressions about music samples recorded into those concert halls.

No description available.

The purpose of this study is to undertake the analysis of underwater detection and tracking using the Doppler phase-shift effects to enhance the detection capability. The fundamental principle is owing to the fact that the M-sequence may result in a better distinction to the echo returning from a moving target than the traditional LFM signal, in that the matched filter using M-sequence may need to estimate and compensate the doppler shift due to the moving target. The experiments were carried out in two harbors: True Love Pier of Kaohsiung Harbor (TLPKH) and Woods Hole Hrabor (WHH). The TLPKH is an inner harbor, with sediment being mud, while the WHH is an open types, suitable for target detection. The results from WHH experiment has shown that when the results from M-sequence and traditional LFM signal were compared, the M-sequence yields much better capability both in detection and estimation of the speed of the moving target along the beam axis. However, the signals from TLPKH were too weak for analysis, therefore, the data from TLPKH were used to analyze the environmental noise, transmission loss, which were combined with estimated values for sonar parameters to conduct the sonar performance analysis in an harborenvironment.

It is widely known that intonation in live professional trombone auditions is one of the most critical factors for which execution is paramount. However, the musician who practices dutifully and precisely with a chromatic tuner, even to the point of technical mastery, will not be prepared sufficiently. He or she will find that in certain environments where heavy reverberation is present, the harmonies inadvertently created are not in tune, even when equal-tempered tuning is executed perfectly, due to the harmonic interactions that those reverberations create. Therefore, it is important that trombonists know how to play auditions excerpts with just intonation, a system that accounts for harmony to deliver results that are truly in tune, for use in the solo round of an audition in such an acoustically "wet" space. This document demonstrates the need for a solution in this regard, the factors involved in a practical application of these concepts in varying scenarios, and presents analyses in just intonation of ten of the most commonly requested excerpts. In addition, guidance and resources are provided for application beyond the excerpts that have been included. It is intended that the trombonist who reads this document will have a better understanding of the basics of just intonation as they apply to solo auditions, so that the quality of his or her audition is improved by leaving at least one less element, intonation, up to chance.

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

Στην παρούσα διπλωματική εργασία γίνεται μελέτη των στατιστικών παραμέτρων ηχητικών σημάτων. Μελετάται η δυνατότητα αυτόματης ταξινόμησης μουσικής ανά είδος, η οποία βασίζεται στην εξαγωγή αυτών των παραμέτρων. Επίσης, μελετάται η μεταβολή αυτών σε συνθήκες αντήχησης, δίνοντας έμφαση στην παράμετρο φασματικής ασυμμετρίας ηχητικού σήματος. Σε αυτό το πλαίσιο, προτείνεται μέθοδος κατασκευής μοντέλου πρόβλεψης της συμπεριφοράς της συγκεκριμένης παραμέτρου σε συνθήκες αντήχησης, που στόχο έχει την εκτίμηση της απόστασης ηχητικής πηγής – δέκτη σε κλειστό χώρο, καθώς και την πρόβλεψη της ανωτέρω παραμέτρου ανηχωικού σήματος από σήματα με αντήχηση. / In this thesis we study the audio features and their applications, such as automatic music genre classification. It is also studied the behavior of these features under reverberant conditions, emphasizing on spectral skewness. In this framework, it is suggested a method of predicting the behavior of this feature under reverberant conditions, which could have many applications such as source - receiver distance estimation and prediction of the spectral skewness of anechoic audio signals.