15 research outputs found

    Binaural to multichannel audio upmix

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    Audion tallennus- ja toistolaitteiden valikoiman kasvaessa on tärkeää, että kaikenlaisilla välineillä tallennettua sekä syntetisoitua audiota voidaan muokata toistettavaksi kaikenlaisilla äänentoistojärjestelmillä. Tässä diplomityössä esitellään menetelmä, jolla binauraalinen audiosignaali voidaan muokata toistettavaksi monikanavaisella kaiutinjärjestelmällä säilyttäen signaalin suuntainformaation. Tällaiselle muokkausmenetelmälle on tarvetta esimerkiksi etäläsnäolosovelluksissa keinona toistaa binauraalinen äänitys monikanavaisella kaiutinjärjestelmällä. Menetelmässä binauraalisesta signaalista estimoidaan ensin äänilähteiden suunnat käyttäen hyväksi korvien välistä aikaeroa. Signaali muokataan monofoniseksi, ja tulosuunnan estimoinnin antama tieto tallennetaan sivuinformaationa. Monofoninen signaali muokataan sen jälkeen halutulle monikanavaiselle kaiutinjärjestelmälle panoroimalla se tallennetun suuntainformaation mukaisesti. Käytännössä menetelmä siis muuntaa korvien välisen aikaeron kanavien väliseksi voimakkuuseroksi. Menetelmässä käytetään ja yhdistellään olemassaolevia tekniikoita tulosuunnan estimoinnille sekä panoroinnille. Menetelmää testattiin vapaamuotoisessa kuuntelukokeessa, sekä lisäämällä ääninäytteisiin binauraalista taustamelua ennen muokkausta ja arvioimalla sen vaikutusta muokatun signaalin laatuun. Menetelmän todettiin toimivan kelvollisesti sekä suuntainformaation säilymisen, että äänen laadun suhteen, ottaen huomioon, että sen kehitystyö on vasta aluillaan.The increasing diversity of popular audio recording and playback systems gives reasons to ensure that recordings made with any equipment, as well as any synthesised audio, can be reproduced for playback with all types of devices. In this thesis, a method is introduced for upmixing binaural audio into a multichannel format while preserving the correct spatial sensation. This type of upmix is required when a binaural recording is desired to be spatially reproduced for playback over a multichannel loudspeaker setup, a scenario typical for e.g. the prospective telepresence appliances. In the upmix method the sound source directions are estimated from the binaural signal by using the interaural time difference. The signal is then downmixed into a monophonic format and the data given by the azimuth estimation is stored as side-information. The monophonic signal is upmixed for an arbitrary multichannel loudspeaker setup by panning it on the basis of the spatial side-information. The method, thus effectively converting interaural time differences into interchannel level differences, employs and conjoins existing techniques for azimuth estimation and discrete panning. The method was tested in an informal listening test, as well as by adding spatial background noise into the samples before upmixing and evaluating its influence on the sound quality of the upmixed samples. The method was found to perform acceptably well in maintaining both the spatiality as well as the sound quality, regarding that much development work remains to be done

    An investigation into the real-time manipulation and control of three-dimensional sound fields

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    This thesis describes a system that can be used for the decoding of a three dimensional audio recording over headphones or two, or more, speakers. A literature review of psychoacoustics and a review (both historical and current) of surround sound systems is carried out. The need for a system which is platform independent is discussed, and the proposal for a system based on an amalgamation of Ambisonics, binaural and transaural reproduction schemes is given. In order for this system to function optimally, each of the three systems rely on providing the listener with the relevant psychoacoustic cues. The conversion from a five speaker ITU array to binaural decode is well documented but pair-wise panning algorithms will not produce the correct lateralisation parameters at the ears of a centrally seated listener. Although Ambisonics has been well researched, no one has, as yet, produced a psychoacoustically optimised decoder for the standard irregular five speaker array as specified by the ITU as the original theory, as proposed by Gerzon and Barton (1992) was produced (known as a Vienna decoder), and example solutions given, before the standard had been decided on. In this work, the original work by Gerzon and Barton (1992) is analysed, and shown to be suboptimal, showing a high/low frequency decoder mismatch due to the method of solving the set of non-linear simultaneous equations. A method, based on the Tabu search algorithm, is applied to the Vienna decoder problem and is shown to provide superior results to those shown by Gerzon and Barton (1992) and is capable of producing multiple solutions to the Vienna decoder problem. During the write up of this report Craven (2003) has shown how 4th order circular harmonics (as used in Ambisonics) can be used to create a frequency independent panning law for the five speaker ITU array, and this report also shows how the Tabu search algorithm can be used to optimise these decoders further. A new method is then demonstrated using the Tabu search algorithm coupled with lateralisation parameters extracted from a binaural simulation of the Ambisonic system to be optimised (as these are the parameters that the Vienna system is approximating). This method can then be altered to take into account head rotations directly which have been shown as an important psychoacoustic parameter in the localisation of a sound source (Spikofski et al., 2001) and is also shown to be useful in differentiating between decoders optimised using the Tabu search form of the Vienna optimisations as no objective measure had been suggested. Optimisations for both Binaural and Transaural reproductions are then discussed so as to maximise the performance of generic HRTF data (i.e. not individualised) using inverse filtering methods, and a technique is shown that minimises the amount of frequency dependant regularisation needed when calculating cross-talk cancellation filters.EPRS

    Modification of multichannel audio for non-standard loudspeaker configurations

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    Tämä diplomityö käsittelee monikanavaäänen analyysi- ja hajotelmamenetelmiä. Työn tavoitteena on pystyä muokkaamaan monikanavaäänityksiä uusille kaiutinkokoonpanoille siten, että äänen tilaominaisuudet säilyvät. Teoriataustana työssä ovat ihmiskuulon tilahavainnointiominaisuudet, äänisignaaleihin perustuvat samankaltaisuusmitat sekä suunta-arviot ja informaatioteknologian lähde-erottelumenetelmät. Työ käy läpi kirjallisuudesta löytyviä monikanavaäänen muokkausmenetelmiä. Diplomityön kokeellisen osuuden aloittaa DVD-levyjen analyysi, jolla pyrittiin saamaan tietoa levyjen äänituotannossa käytettävistä menetelmistä myöhempää äänimuunnostekniikoiden kehittämistä varten. Koe osoitti, että kolmen etukanavasignaalin ja kahden takakanavasignaalin välillä on vain harvoin yhteisiä äänikomponentteja. Kompaktien kaiutinkokoonpanojen ominaisuuksia tutkittiin kahdessa kuuntelukokeessa. Ensimmäinen koe tarkasteli eroja eri kolmikanavaisten kaiutinasettelujen välillä. Tavoitteena näissä toistosysteemeissä oli hyödyntää ääniaaltojen heijastuksia huoneen seinistä. Jälkimmäinen kuuntelukoe sovelsi kolmea tunnettua äänimuunnosmenetelmää kolmikanavaiseen kompaktiin kaiutinkokoonpanoon, jonka toistosta saatavaa tilahavaintoa pyrittiin laajentamaan. Kahden metodeista havaittiin parantavan tutkittuja tilaominaisuuksia.In this thesis, analysis and decomposition methods for multichannel audio are studied. The objective of the work is to transform multichannel recordings to new reproduction systems so that the spatial properties of the sound are preserved. Spatial hearing of the human auditory system, signal-based similarity and localization measures, and information-technological source separation methods are described as background theory. Then, different multichannel audio transform methods are reviewed. The experimental part of the work starts with an analysis of DVD recordings to gain helpful information about the production methods of such recordings for further development of audio transform methods. The test reveals that the three frontal channels do not usually share common sound sources with the two rear channels. The properties of compact loudspeaker systems are investigated in two listening tests. The first test studies the differences between three-channel loudspeaker layouts, which exploit the reflections of sound waves from room boundaries. The latter one of the tests applies three transform methods known from the literature to widen the spatial dimensions of a three-channel compact loudspeaker system in comparison to a reference stereo system. These methods are a stereo signal transform method based on signal powers and interchannel cross-correlations, a primaryambient signal decomposition based on principal component analysis (PCA), and directional audio coding (DirAC). The methods were ranked in this descending order of preference by the test subjects

    Subjective evaluation and electroacoustic theoretical validation of a new approach to audio upmixing

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    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

    Evaluating the Perceived Quality of Binaural Technology

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    This thesis studies binaural sound reproduction from both a technical and a perceptual perspective, with the aim of improving the headphone listening experience for entertainment media audiences. A detailed review is presented of the relevant binaural technology and of the concepts and methods for evaluating perceived quality. A pilot study assesses the application of state-of-the-art binaural rendering systems to existing broadcast programmes, finding no substantial improvements in quality over conventional stereo signals. A second study gives evidence that realistic binaural simulation can be achieved without personalised acoustic calibration, showing promise for the application of binaural technology. Flexible technical apparatus is presented to allow further investigation of rendering techniques and content production processes. Two web-based studies show that appropriate combination of techniques can lead to improved experience for typical audience members, compared to stereo signals, even without personalised rendering or listener head-tracking. Recent developments in spatial audio applications are then discussed. These have made dynamic client-side binaural rendering with listener head-tracking feasible for mass audiences, but also present technical constraints. To limit distribution bandwidth and computational complexity during rendering, loudspeaker virtualisation is widely used. The effects on perceived quality of these techniques are studied in depth for the first time. A descriptive analysis experiment demonstrates that loudspeaker virtualisation during binaural rendering causes degradations to a range of perceptual characteristics and that these vary across other system conditions. A final experiment makes novel use of the check-all-that-apply method to efficiently characterise the quality of seven spatial audio representations and associated dynamic binaural rendering techniques, using single sound sources and complex dramatic scenes. The perceived quality of these different representations varies significantly across a wide range of characteristics and with programme material. These methods and findings can be used to improve the quality of current binaural technology applications

    Simulation and analysis of spatial audio reproduction and listening area effects

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    Loudspeaker-based spatial audio systems are often designed with the aim to create an auditory event or scene to a listener positioned in the optimal listening position. However, in real-world domestic listening environments, listeners can be distributed across the listening area. Any translational change from the central listening position will introduce artefacts which can be challenging to evaluate perceptually. Simulation of a loudspeaker system using non-individualised dynamic binaural synthesis is one solution to this problem. However, the validity in using such systems is not well proven.This thesis measures the limitations of using a non-individualised, dynamic binaural synthesis system to simulate the perception of loudspeaker-based panning methods across the listening area. The binaural simulation system was designed and verified in collaboration with BBC Research and Development. The equivalence of localisation errors caused by loudspeaker-based panning methods between in situ and binaural simulation was measured where it was found that localisation errors were equivalent to a +/-7 degrees boundary in 75% of the spatial audio reproduction systems tested. Results were then compared to a computation localisation model which was adapted to utilise head-rotations. The equivalence of human acuity to sound colouration between in situ and when using non-individualised binaural simulation was measured using colouration detection thresholds from five directions. It was shown that thresholds were equivalent within a +/-4dB equivalence boundary, supporting the use for simulating sound colourations caused by loudspeaker-based panning methods. The binaural system was finally applied to measure the perception of multi-loudspeaker induced colouration artefacts across the listening area. It was found that the central listening position had the lowest perceived colouration. It is also shown that the variation in perceived colouration across the listening area is larger for reverberant reproduction conditions

    Object-based radio : effects on production and audience experience

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    This thesis analyses the benefits of using object-based audio as a production and delivery format in order to enable new audience experiences. This is achieved though a series of case studies, each focusing on a different user experience enabled by the use of object-based audio. Each study considers the impact of using object-based audio on the creative process, production workflow and audience experience.The first study analyses the audience’s use of the ability to personalise the mix of a live football match. It demonstrates that there was not a single audio mix favoured by all, and the ability to change the mix was valued by the audience. While listeners did adjust the mix initially, they tended to leave it at that setting and did not interact much once they made their initial selection. While there were three favoured mixes, over 50% of listeners did not choose one of these three mixes, indicating that only offering three options would not satisfy everyone.Modes of listening model the ways listeners deconstruct complex sound scenes into foreground and background categories ascribing different salience to foreground and background sounds. The second study uses this model to inform a series of card sorting exercises which result in similar foreground and background categories. However, rather than being unimportant, background sounds were present to convey ancillary information or to affect emotional responses and foreground sounds to expose plot or story events. This study demonstrated that this grouping was a meaningful categorisation for broadcast sound and evaluated how beneficial allowing different foreground and background audio mixes would be for audiences. It contains analysis of audio objects in the context of foreground and background sounds based on the opinions of the content creators. It also includes subjective testing of audience preferences for different mixes of foreground verses background audio levels across five different genres and four different loudspeaker layouts. It shows that there is no clustering of listeners based on their preference of foreground vs background balances. It also shows that there is significant variation of foreground and background balance preference between loudspeaker layouts.The final study goes beyond tailoring audio levels, balances and loudspeaker layouts and analyses the benefit to audiences of being able to adapt the story of a drama in order to set it in a location that is familiar to the listener. It shows that being able to set a radio drama in the location where the listening is taking place improves audience’s enjoyment of the programme. 75% of listeners who experienced the tailored version of the drama reported liking the story, compared with 65% of listeners who experienced a non-tailored version.The three studies also analyse the impact of object-based content creation on production workflows by documenting the challenges faced and discussing possible solutions. For example, providing writers with constraints when they are designing dynamic content and allowing sound designers time to develop trust in the technology when mixing content for multiple loudspeaker layouts.The original contribution to knowledge is to establish a new listening model applicable to constructed and designed sound experiences based on functional analysis of audio objects. This work also establishes, for the first time, a framework for the definition of an audio object based on the creator’s intended range of audience experiences. In addition the thesis also provides insights into how audiences interact with object-based content experiences and insights about audience attitudes towards using personal data to personalise object-based content experiences. Each study addresses the potential advantages of delivering object-based audio, assess any impact on the quality of the audience’s experience and analyses the challenges faced by production in the creation of these new experiences

    Perceptual evaluation of personal, location-aware spatial audio

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    This thesis entails an analysis, synthesis and evaluation of the medium of personal, location aware spatial audio (PLASA). The PLASA medium is a specialisation of locative audio—the presentation of audio in relation to the listener’s position. It also intersects with audio augmented reality—the presentation of a virtual audio reality, superimposed on the real world. A PLASA system delivers binaural (personal) spa- tial audio to mobile listeners, with body-position and head-orientation interactivity, so that simulated sound source positions seem fixed in the world reference frame. PLASA technical requirements were analysed and three system architectures identified, employing mobile, remote or distributed rendering. Knowledge of human spatial hearing was reviewed to ascertain likely perceptual effects of the unique factors of PLASA compared to static spatial audio. Human factors identified were multimodal perception of body-motion interaction and coincident visual stimuli. Technical limitations identified were rendering method, individual binaural rendering, and accuracy and latency of position- and orientation-tracking. An experimental PLASA system was built and evaluated technically, then four perceptual experiments were conducted to investigate task-related perceptual per- formance. These experiments tested the identified human factors and technical limitations against performance measures related to localisation and navigation tasks, under conditions designed to be ecologically valid to PLASA application scenarios. A final experiment assessed navigation task performance with real sound sources and un-mediated spatial hearing for comparison with virtual source performance. Results found that body-motion interaction facilitated correction of front–back confusions. Body-motion and the multi-modal stimuli of virtual–audible and real–visible objects supported lower azimuth errors than stationary, mono-modal localisation of the same audio-only stimuli. PLASA users navigated efficiently to stationary virtual sources, despite varied rendering quality and head-turn latencies between 176 ms and 976 ms. Factors of rendering method, individualisation and head-turn latency showed interaction effects such as greater sensitivity to latency for some rendering methods than others. In general, PLASA task performance levels agreed with expectations from static or technical performance tests, and some results demonstrated similar performance levels to those achieved in the real-source baseline test
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