Voice over the Internet Protocol (VoIP) is one of the latest and most successful Internet services. It takes advantage of Wireless Local Area Networks (WLANs) and broadband connections to provide high quality and low cost telephony over the Internet or an intranet. This project exploits features of VoIP to create a communication scenario where various conversations can be held at the same time, and each of these conversations can be located at a virtual location in space. The report includes theoretical analysis of psychoacoustic parameters and their experimental implementation together with the design of a spatial audio module for the Session Initiation Protocol (SIP) User Agent “minisip”. Besides the 3D sound environment this project introduces multitasking as an integrative feature for “minisip”, gathering various sound inputs connected by a SIP session to the “minisip” interface, and combining them altogether into a unique output. This later feature is achieved with the use of resampling as a core technology. The effects of traffic increment to and from the user due to the support of multiple streams are also introduced. / Röst över Internet Protocol (VoIP) är en av de senaste och mest framgångsrika Internettjänsterna. Det utnyttjar Trådlösa Nätverk och bredband för att erbjuda högkvalitativ och billig telefonering över Internet eller ett Intranät. Det här projektet använder sig av VoIP för att skapa ett kommunikationsscenario där flera olika konversationer kan hållas samtidigt och där varje konversation kan placeras på en virtuell plats i rymden. Rapporten innehåller en teoretisk analys av psykoakustiska parametrar och deras experimentella genomförande tillsammans med design av en 3D ljud modul för Session Initiation Protocol (SIP) User Agent ”minisip”. Förutom ljudmiljön i 3D introducerar projektet multitasking som en integrerbar del av ”minisip”. Alla tänkbara ljudkällor baserade på SIP förbindelser samlas med ”minisip” interfacet och kombineras till en enda utsignal. Detta uppnås med hjälp av resampling som kärnteknologi. Effekterna av att mer trafik når användaren på grund av stödet av multiple källor introduceras också.

Designing auditory interfaces is a challenge for current human-systems developers. This is largely due to a lack of theoretical guidance for directing how best to use sounds in today's visually-rich graphical user interfaces. This dissertation provided a framework for guiding the design of audio interfaces to enhance human-systems performance. This doctoral research involved reviewing the literature on conveying temporal and spatial information using audio, using this knowledge to build three theoretical models to aid the design of auditory interfaces, and empirically validating select components of the models. The three models included an audio integration model that outlines an end-to-end process for adding sounds to interactive interfaces, a temporal audio model that provides a framework for guiding the timing for integration of these sounds to meet human performance objectives, and a spatial audio model that provides a framework for adding spatialization cues to interface sounds. Each model is coupled with a set of design guidelines theorized from the literature, thus combined, the developed models put forward a structured process for integrating sounds in interactive interfaces. The developed models were subjected to a three phase validation process that included review by Subject Matter Experts (SMEs) to assess the face validity of the developed models and two empirical studies. For the SME review, which assessed the utility of the developed models and identified opportunities for improvement, a panel of three audio experts was selected to respond to a Strengths, Weaknesses, Opportunities, and Threats (SWOT) validation questionnaire. Based on the SWOT analysis, the main strengths of the models included that they provide a systematic approach to auditory display design and that they integrate a wide variety of knowledge sources in a concise manner. The main weaknesses of the models included the lack of a structured process for amending the models with new principles, some branches were not considered parallel or completely distinct, and lack of guidance on selecting interface sounds. The main opportunity identified by the experts was the ability of the models to provide a seminal body of knowledge that can be used for building and validating auditory display designs. The main threats identified by the experts were that users may not know where to start and end with each model, the models may not provide comprehensive coverage of all uses of auditory displays, and the models may act as a restrictive influence on designers or they may be used inappropriately. Based on the SWOT analysis results, several changes were made to the models prior to the empirical studies. Two empirical evaluation studies were conducted to test the theorized design principles derived from the revised models. The first study focused on assessing the utility of audio cues to train a temporal pacing task and the second study combined both temporal (i.e., pace) and spatial audio information, with a focus on examining integration issues. In the pace study, there were four different auditory conditions used for training pace: 1) a metronome, 2) non-spatial auditory earcons, 3) a spatialized auditory earcon, and 4) no audio cues for pace training. Sixty-eight people participated in the study. A pre- post between subjects experimental design was used, with eight training trials. The measure used for assessing pace performance was the average deviation from a predetermined desired pace. The results demonstrated that a metronome was not effective in training participants to maintain a desired pace, while, spatial and non-spatial earcons were effective strategies for pace training. Moreover, an examination of post-training performance as compared to pre-training suggested some transfer of learning. Design guidelines were extracted for integrating auditory cues for pace training tasks in virtual environments. In the second empirical study, combined temporal (pacing) and spatial (location of entities within the environment) information were presented. There were three different spatialization conditions used: 1) high fidelity using subjective selection of a "best-fit" head related transfer function, 2) low fidelity using a generalized head-related transfer function, and 3) no spatialization. A pre- post between subjects experimental design was used, with eight training trials. The performance measures were average deviation from desired pace and time and accuracy to complete the task. The results of the second study demonstrated that temporal, non-spatial auditory cues were effective in influencing pace while other cues were present. On the other hand, spatialized auditory cues did not result in significantly faster task completion. Based on these results, a set of design guidelines was proposed that can be used to direct the integration of spatial and temporal auditory cues for supporting training tasks in virtual environments. Taken together, the developed models and the associated guidelines provided a theoretical foundation from which to direct user-centered design of auditory interfaces.

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Research investigating the behavioral and physiological responses of bats to echoes typically includes analysis of acoustic signals from microphones and/or microphone arrays, using time difference of arrival (TDOA) between array elements or the microphones to locate flying bats (azimuth and elevation). This has provided insight into transmission adaptations with respect to target distance, clutter, and interference. Microphones recording transmitted signals and echoes near a stationary bat provide sound pressure as a function of time but no directional information. This dissertation introduces spatial audio techniques to bat biosonar studies as a complementary method to the current TDOA based acoustical study methods. This work proposes a couple of feasible methods based on spatial audio techniques, that both track bats in flight and pinpoint the directions of echoes received by a bat. A spatial audio/soundfield microphone array is introduced to measure sounds in the sonar frequency range (20-80 kHz) of the big brown bat (Eptesicus fuscus). The custom-built ultrasonic tetrahedral soundfield microphone consists of four capacitive microphones that were calibrated to match magnitude and phase responses using a transfer function approach. Ambisonics, a signal processing technique used in three-dimensional (3D) audio applications, is used for the basic processing and reproduction of the signals measured by the soundfield microphone. Ambisonics provides syntheses and decompositions of a signal containing its directional properties, using the relationship between the spherical harmonics and the directional properties. As the first proposed method, a spatial audio decoding technique called HARPEx (High Angular Resolution Planewave Expansion) was used to build a system providing angle and elevation estimates. HARPEx can estimate the direction of arrivals (DOA) for up to two simultaneous sources since it decomposes a signal into two dominant planewaves. Experiments proved that the estimation system based on HARPEx provides accurate DOA estimates of static or moving sources. It also reconstructed a smooth flight-path of a bat by accurately estimating its direction at each snapshot of pulse measurements in time. The performance of the system was also assessed using statistical analyses of simulations. A signal model was built to generate microphone capsule responses to a virtual source emitting an LFM signal (3 ms, two harmonics: 40-22 kHz and 80-44 kHz) at an angle of 30° in the simulations. Medians and RMSEs (root-mean-square error) of 10,000 simulations for each case represent the accuracy and precision of the estimations, respectively. Results show lower d (distance between a capsule and the soundfield microphone center) or/and higher SNR (signal-to-noise ratio) are required to achieve higher estimator performance. The Cramer-Rao lower bounds (CRLB) of the estimator are also computed with various d and SNR conditions. The CRLB which is for TDOA based methods does not cover the effects of different incident angles to the capsules and signal delays between the capsules due to a non-zero d, on the estimation system. This shows the CRLB is not a proper tool to assess the estimator performance. For the second proposed method, the matched-filter technique is used instead of HARPEx to build another estimation system. The signal processing algorithm based on Ambisonics and the matched-filter approach reproduces a measured signal in various directions, and computes matched-filter responses of the reproduced signals in time-series. The matched-filter result points a target(s) by the highest filter response. This is a sonar-like estimation system that provides information of the target (range, direction, and velocity) using sonar fundamentals. Experiments using a loudspeaker (emitter) and an artificial or natural target (either stationary or moving) show the system provides accurate estimates of the target's direction and range. Simulations of imitating a situation where a bat emits a pulse and receives an echo from a target (30°) were also performed. The echo sound level is determined using the sonar equation. The system processed the virtual bat pulse and echo, and accurately estimated the direction, range, and velocity of the target. The simulation results also appear to recommend an echo level over -3 dB for accurate and precise estimations (below 15% RMSE for all parameters). This work proposes two methods to track bats in flight or/and pinpoint the directions of targets using spatial audio techniques. The suggested methods provide accurate estimates of the direction, range, or/and velocity of a bat based on its pulses or of a target based on echoes. This demonstrates these methods can be used as key tools to reconstruct bat biosonar. They would be also an independent tool or a complementary option to TDOA based methods, for bat echolocation studies. The developed methods are believed to be also useful in improving man-made sonar technology. / Doctor of Philosophy / While bats are one of the most intriguing creatures to the general population, they are also a popular subject of study in various disciplines. Their extraordinary ability to navigate and forage irrespective of clutter using echolocation has gotten attention from many scientists and engineers. Research investigating bats typically includes analysis of acoustic signals from microphones and/or microphone arrays. Using time difference of arrival (TDOA) between the array elements or the microphones is probably the most popular method to locate flying bats (azimuth and elevation). Microphone responses to transmitted signals and echoes near a bat provide sound pressure but no directional information. This dissertation proposes a complementary way to the current TDOA methods, that delivers directional information by introducing spatial audio techniques. This work shows a couple of feasible methods based on spatial audio techniques, that can both track bats in flight and pinpoint the directions of echoes received by a bat. An ultrasonic tetrahedral soundfield microphone is introduced as a measurement tool for sounds in the sonar frequency range (20-80 kHz) of the big brown bat (Eptesicus fuscus). Ambisonics, a signal processing technique used in three-dimensional (3D) audio applications, is used for the basic processing of the signals measured by the soundfield microphone. Ambisonics also reproduces a measured signal containing its directional properties. As the first method, a spatial audio decoding technique called HARPEx (High Angular Resolution Planewave Expansion) was used to build a system providing angle and elevation estimates. HARPEx can estimate the direction of arrivals (DOA) for up to two simultaneous sound sources. Experiments proved that the estimation system based on HARPEx provides accurate DOA estimates of static or moving sources. The performance of the system was also assessed using statistical analyses of simulations. Medians and RMSEs (root-mean-square error) of 10,000 simulations for each simulation case represent the accuracy and precision of the estimations, respectively. Results show shorter distance between a capsule and the soundfield microphone center, or/and higher SNR (signal-to-noise ratio) are required to achieve higher performance. For the second method, the matched-filter technique is used to build another estimation system. This is a sonar-like estimation system that provides information of the target (range, direction, and velocity) using matched-filter responses and sonar fundamentals. Experiments using a loudspeaker (emitter) and an artificial or natural target (either stationary or moving) show the system provides accurate estimates of the target's direction and range. Simulations imitating a situation where a bat emits a pulse and receives an echo from a target (30°) were also performed. The system processed the virtual bat pulse and echo, and accurately estimated the direction, range, and velocity of the target. The suggested methods provide accurate estimates of the direction, range, or/and velocity of a bat based on its pulses or of a target based on echoes. This demonstrates these methods can be used as key tools to reconstruct bat biosonar. They would be also an independent tool or a complementary option to TDOA based methods, for bat echolocation studies. The developed methods are also believed to be useful in improving sonar technology.

This thesis is related to the field of Sound Source Separation (SSS). It addresses the development and evaluation of these techniques for their application in the resynthesis of high-realism sound scenes by means of Wave Field Synthesis (WFS). Because the vast majority of audio recordings are preserved in twochannel stereo format, special up-converters are required to use advanced spatial audio reproduction formats, such as WFS. This is due to the fact that WFS needs the original source signals to be available, in order to accurately synthesize the acoustic field inside an extended listening area. Thus, an object-based mixing is required. Source separation problems in digital signal processing are those in which several signals have been mixed together and the objective is to find out what the original signals were. Therefore, SSS algorithms can be applied to existing two-channel mixtures to extract the different objects that compose the stereo scene. Unfortunately, most stereo mixtures are underdetermined, i.e., there are more sound sources than audio channels. This condition makes the SSS problem especially difficult and stronger assumptions have to be taken, often related to the sparsity of the sources under some signal transformation. This thesis is focused on the application of SSS techniques to the spatial sound reproduction field. As a result, its contributions can be categorized within these two areas. First, two underdetermined SSS methods are proposed to deal efficiently with the separation of stereo sound mixtures. These techniques are based on a multi-level thresholding segmentation approach, which enables to perform a fast and unsupervised separation of sound sources in the time-frequency domain. Although both techniques rely on the same clustering type, the features considered by each of them are related to different localization cues that enable to perform separation of either instantaneous or real mixtures.Additionally, two post-processing techniques aimed at improving the isolation of the separated sources are proposed. The performance achieved by several SSS methods in the resynthesis of WFS sound scenes is afterwards evaluated by means of listening tests, paying special attention to the change observed in the perceived spatial attributes. Although the estimated sources are distorted versions of the original ones, the masking effects involved in their spatial remixing make artifacts less perceptible, which improves the overall assessed quality. Finally, some novel developments related to the application of time-frequency processing to source localization and enhanced sound reproduction are presented. / Cobos Serrano, M. (2009). Application of sound source separation methods to advanced spatial audio systems [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/8969

Résumé : La synthèse de champs sonores est un domaine de recherche actif trouvant de nombreuses applications musicales, multimédias ou encore industrielles. Dans ce dernier cas, la re- construction précise du champ sonore est souhaitée, ce qui implique de répondre à un certains nombre de questionnements scientifiques. À l’aide de réseaux de microphones et de haut-parleurs, la captation, la synthèse et la reconstruction précise de champs sonores sont théoriquement possibles. Seulement, pour des applications pratiques, la disposition des haut-parleurs et l’influence acoustique du lieu de restitution sont des facteurs cruciaux à prendre en compte pour s’assurer de la bonne reconstruction du champ sonore. Dans ce contexte, cette thèse de doctorat propose des méthodes et des techniques pour la captation, la transformation et la reconstruction précise de champs sonores en trois dimen- sions en se basant sur la méthode ambisonique d’ordre élevé. Une configuration sphérique pour le réseau de microphones et de haut-parleurs est proposée. Elle suit un maillage de Lebedev à cinquante points qui permet la captation et la reconstruction du champ sonore jusqu’à l’ordre 5 avec le formalisme ambisonique. Les limitations de cette approche, tel le repliement spatial, sont étudiés en détails. De plus, une opération de transformation du champ sonore est présentée. Elle est établie dans le domaine des harmoniques sphériques et permet d’effectuer un filtrage directionnel avant le décodage pour privilégier certaines directions dans le champ sonore, suivant une fonction de directivité choisie. Pour la re- construction, une approche originale, également établie dans le domaine des harmoniques sphériques, permet de prendre en compte l’influence acoustique du lieu de restitution, ainsi que les défauts du système de restitution. Ce traitement permet alors d’adapter la synthèse de champs sonores au lieu de restitution, en conservant le formalisme théorique établi en champ libre. Finalement, une validation expérimentale des méthodes et des tech- niques développées au cours de la thèse est faite. Dans ce contexte, une suite logicielle de synthèse et traitement en temps-réel des champs sonore est développée. / Abstract : Sound field synthesis is an active research domain with various musical, multimedia or industrial applications. In the latter case, the accurate reconstruction of the sound field is targeted, which involves answering several scientific questions. Using arrays of microphones and loudspeakers, the capture, synthesis and accurate reconstruction of sound fields are theoretically possible. However, for practical applications, the arrangement of the loud- speakers and the acoustic influence of the restitution room are critical factors to consider in order to ensure the accurate reconstruction of the sound field. In this context, this thesis proposes methods and techniques for the capture, transforma- tions and accurate reconstruction of sound fields in three dimensions based on the Higher Order Ambisonics (HOA) method. A spherical configuration for the array of microphones and loudspeakers is proposed. It follows a fifty-node Lebedev grid that enables the capture and reconstruction of the sound field up to order 5 with HOA formalism. The limitations of this approach, such as the spatial aliasing, are studied in detail. A transformation op- eration of the sound field is also proposed. The formulation is established in the spherical harmonics domain and enables a directional filtering on the sound field prior to the decod- ing step. For the reconstruction of the sound field, an original approach, also established in the spherical harmonics domain, can take into account the acoustic influence of the restitution room and the defects of the playback system. This treatment then adapts the synthesis of sound fields to the restitution room, maintaining the theoretical formalism established in free field. Finally, an experimental validation of methods and techniques developed in the thesis is made. In this context, a digital signal processing toolkit is de- veloped. It process in real-time the microphones, ambisonics, and loudspeaker signals for the sound field capture, transformations, and decoding.

The increasing interest for immersive experiences in areas such as augmented and virtual reality makes high quality 3D sound more important than ever before. A technique for capturing and rendering 3D audio which has received more attention during the last twenty years are Higher Order Ambisonics (HOA). Higher Order Ambisonics is a scene based audio format which has a lot of advantages compared to other standard formats. Hovever, one problem with HOA is that it requires a lot of bandwidth. For example, sending an uncoded high quality HOA signal requires 49 channels to be transmitted at the same time which requires a bandwidth of about 40 Mbps. A lot of effort has been made in the last ten years on coding HOA signals. In this thesis, two different approaches are taken on coding HOA signals. In one approach, called Sound Field Rotation (SFR) in this thesis, the microphone that records the sound field is virtually rotated to see if it is possible to make some of the channels zero. The second approach, called Sound Field Decomposition (SFD) in this thesis, use Principal component analysis to decompose a sound field into a foreground and background component. The Sound Field Decomposition approach is inspired by the emerging MPEG-H 3D Audio standard for coding HOA signals. The result shows that the Sound Field Rotation method only works for very simple sound scenes. It has also been shown that a 49 channels HOA signal can be reduced to as little as 7 channels if the sound scene consists of a point source. The Sound Field Deomposition method worked for more complex sound scenes. It was shown that a MPEG similar system could be improved. Result from MUSHRA (Multiple stimuli with hidden reference and anchor) listening tests showed that an improved MPEG similar system reached a MUSHRA score about 78 while the MPEG similar system reached 55 at a bitrate of 256 kbps. Without coding each monochannels with the 3GPP EVS (Enhanced voice services) codec, the improved MPEG similar system reached the MUSHRA score 85. At 256 kbps, the improved MPEG similar system coded the HOA signal into six channels instead of 49 for the uncoded signal. From objective results, it was shown that the improved MPEG similar system had largest effect at low bitrates.

abstract: Several music players have evolved in multi-dimensional and surround sound systems. The audio players are implemented as software applications for different audio hardware systems. Digital formats and wireless networks allow for audio content to be readily accessible on smart networked devices. Therefore, different audio output platforms ranging from multispeaker high-end surround systems to single unit Bluetooth speakers have been developed. A large body of research has been carried out in audio processing, beamforming, sound fields etc. and new formats are developed to create realistic audio experiences. An emerging trend is seen towards high definition AV systems, virtual reality gears as well as gaming applications with multidimensional audio. Next generation media technology is concentrating around Virtual reality experience and devices. It has applications not only in gaming but all other fields including medical, entertainment, engineering, and education. All such systems also require realistic audio corresponding with the visuals. In the project presented in this thesis, a new portable audio hardware system is designed and developed along with a dedicated mobile android application to render immersive surround sound experiences with real-time audio effects. The tablet and mobile phone allow the user to control or “play” with sound directionality and implement various audio effects including sound rotation, spatialization, and other immersive experiences. The thesis describes the hardware and software design, provides the theory of the sound effects, and presents demonstrations of the sound application that was created. / Dissertation/Thesis / Masters Thesis Electrical Engineering 2016

This paper discusses technical and conceptual approaches to incorporate 3D spatial movement in electroacoustic music. The Ambisonic spatial audio format attempts to recreate a full sound field (with height information) and is currently a popular choice for 3D spatialization. While tools for Ambisonics are typically designed for the 2D computer screen and keyboard/mouse, virtual reality offers new opportunities to work with spatial audio in a 3D computer generated environment. An overview of my custom virtual reality software, VRSoMa, demonstrates new possibilities for the design of 3D audio. Created in the Unity video game engine for use with the HTC Vive virtual reality system, VRSoMa utilizes the Google Resonance SDK for spatialization. The software gives users the ability to control the spatial movement of sound objects by manual positioning, a waypoint system, animation triggering, or through gravity simulations. Performances can be rendered into an Ambisonic file for use in digital audio workstations. My work Discords (2018) for 3D audio facilitates discussion of the conceptual and technical aspects of spatial audio for use in musical composition. This includes consideration of human spatial hearing, technical tools, spatial allusion/illusion, and blending virtual/real spaces. The concept of spatial gestures has been used to categorize the various uses of spatial motion within a musical composition.