Audio signal processing systems for converting two-channel (stereo) recordings to four or five channels are increasingly relevant. These audio upmixers can be used with conventional stereo sound recordings and reproduced with multichannel home theatre or automotive loudspeaker audio systems to create a more engaging and natural-sounding listening experience. This dissertation discusses existing approaches to audio upmixing for recordings of musical performances and presents specific design criteria for a system to enhance spatial sound quality. A new upmixing system is proposed and evaluated according to these criteria and a theoretical model for its behavior is validated using empirical measurements. / The new system removes short-term correlated components from two electronic audio signals using a pair of adaptive filters, updated according to a frequency domain implementation of the normalized-least-means-square algorithm. The major difference of the new system with all extant audio upmixers is that unsupervised time-alignment of the input signals (typically, by up to +/-10 ms) as a function of frequency (typically, using a 1024-band equalizer) is accomplished due to the non-minimum phase adaptive filter. Two new signals are created from the weighted difference of the inputs, and are then radiated with two loudspeakers behind the listener. According to the consensus in the literature on the effect of interaural correlation on auditory image formation, the self-orthogonalizing properties of the algorithm ensure minimal distortion of the frontal source imagery and natural-sounding, enveloping reverberance (ambiance) imagery. / Performance evaluation of the new upmix system was accomplished in two ways: Firstly, using empirical electroacoustic measurements which validate a theoretical model of the system; and secondly, with formal listening tests which investigated auditory spatial imagery with a graphical mapping tool and a preference experiment. Both electroacoustic and subjective methods investigated system performance with a variety of test stimuli for solo musical performances reproduced using a loudspeaker in an orchestral concert-hall and recorded using different microphone techniques. / The objective and subjective evaluations combined with a comparative study with two commercial systems demonstrate that the proposed system provides a new, computationally practical, high sound quality solution to upmixing.

This thesis investigates the feasibility of using gestures as a means of control for localizing three dimensional (3D) sound sources in a distributed immersive audio system. A prototype system was implemented and tested which uses state of the art technology to achieve the stated goals. A Windows Kinect is used for gesture recognition which translates human gestures into control messages by the prototype system, which in turn performs actions based on the recognized gestures. The term distributed in the context of this system refers to the audio processing capacity. The prototype system partitions and allocates the processing load between a number of endpoints. The reallocated processing load consists of the mixing of audio samples according to a specification. The endpoints used in this research are XMOS AVB endpoints. The firmware on these endpoints were modified to include the audio mixing capability which was controlled by a state of the art audio distribution networking standard, Ethernet AVB. The hardware used for the implementation of the prototype system is relatively cost efficient in comparison to professional audio hardware, and is also commercially available for end users. the successful implementation and results from user testing of the prototype system demonstrates how it is a feasible option for recording the localization of a sound source. The ability to partition the processing provides a modular approach to building immersive sound systems. This removes the constraint of a centralized mixing console with a predetermined speaker configuration.

The thesis deal with the design and construction of single cluster of apparatus which can serve as a signal source for wavefield synthesis. Apparatus can be controlled remotely by Ethernet interface. Thesis contains theoretical analyses of speaker system, class D amplifier, digital to analog conversion, digital format of acoustic data ADAT and Ethernet interface. Conceptual design of cluster and description of constructed module is concerned.

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This thesis investigates the role of recording techniques in immersive music production, aiming to balance practicality and necessity. Through four case studies involving harp, piano, sitar and an original music production inspired by a letter, the research examines how the creative process within music production and recording techniques are influenced by immersive music in mind. Utilizing qualitative observation and quantitative technical experiments, the study documents the process and outcomes, revealing the significance of immersive microphone techniques for maintaining realistic spatialization. Future directions include further exploration of recording techniques and creative applications within immersive music production. By comparing various recording methods, this work contributes a unique perspective to the field, enhancing our understanding of the interplay between technology, creativity, and musical immersion. / <p><strong>Harp: </strong>Improvised piece played by Pauline Burke-Clason. Files: Harp_ADM_Mix1.wav, Harp_Binaural _Mix1.wav,Harp_ADM_Mix2.wav, Harp_Binaural _Mix2.wav and Harp_StereoReference.wav.</p><p>Shorter versions of these files are available on DiVa, while the complete files are stored at the Royal College of Music in Stockholm.</p><p><strong>Grand Piano: </strong><em>Maurice Ravel - Miroirs, No.5, La vallée des cloches</em>, played by Herdís Ágústa Linnet. Files:MauriceRavel-VLaValleeDesCloches_ADM_Mix1.wav, MauriceRavel-VLaValleeDesCloches_Binaural _Mix1.wav, MauriceRavel-VLaValleeDesCloches_ADM_Mix2.wav, MauriceRavel-VLaValleeDesCloches_Binaural_Mix2.wav and MauriceRavel-VLaValleeDesCloches_StereoReference.wav.</p><p>Shorter versions of these files are available on DiVa, while the complete files are stored at the Royal College of Music in Stockholm.</p><p><strong>Sitar: </strong>Improvised piece played by Arjun K. Verma. Files: Sitar_ADM.wav, Sitar_Binaural.wav och Sitar_StereoReference.wav.</p><p>21 seconds of this piece is used in this project.</p><p><strong>The Dream: </strong>Song written and produced by Basma Jabbar.</p><p>Acoustic guitar arranged by and played by Tom Jondelius.</p><p>Electric guitar arranged by and played by Chris Lee.</p><p>Drums played by Felix Kling.</p><p>Strings by Stockholm Studio Orchestra:</p><p>Violin: Anna Roos Stefansson, Claudia Bonfiglioli, Danial Shariati, Jannika Gustafsson, Jonna Simonsson, Lola Torrente, Oscar Treitler, Paul Waltman, Sarah Cross, Simona Bonfiglioli, Vicky Sayles.</p><p>Viola: Christopher Öhman, Erik Holm, Shahar Rosenthal, Vidar Andersson Meilink.</p><p>Cello: Daniel Thorell, Filip Lundberg, Josef Alin.</p><p>Double Bass: Bård Ericson.</p><p>Strings arranged by Henrik Langemyr.</p><p>Strings conducted by Erik Arvinder.</p><p>Strings recorded &amp; engineered by Willem Bleeker &amp; Erik Arvinder, assisted by Basma Jabbar.</p><p>Files: TheDream_ADM.wav, TheDream_Binaural.wav and TheDream_StereoReference.wav.</p><p>Shorter versions of these files are available on DiVa, while the complete files are stored at the Royal College of Music in Stockholm.</p><p>If nothing else is stated, recording and mixing engineer is Basma Jabbar.</p>

Virtual three-dimensional auditory displays (V3DADs) use digital signal processing to deliver sounds (typically through headphones) that seem to originate from specific external spatial locations. This set of studies investigates the delivery of V3DADs through bone-conduction transducers (BCTs) in addition to conventional headphones. Although previous research has shown that spatial separation can be induced through BCTs, some additional signal adjustments are required for optimization of V3DADs, due to the difference in hearing pathways. The present studies tested a bone-conduction adjustment function (BAF) derived from equal-loudness judgments on pure tones whose frequencies were spaced one critical band apart. Localization performance was assessed through conventional air-conduction headphones, BCTs with only transducer correction, and BCTs with a BAF. The results showed that in the elevation plane, the BAF was effective in restoring the spectral cues altered by the bone-conduction pathway. No evidence for increased percept variability or decreased lateralization in the bone-conduction conditions was found. These findings indicate that a V3DAD can be implemented on a BCT and that a BAF will improve performance, but that there is an apparent performance cost that cannot be addressed with BAFs measured using the methodology in the present studies.