Bulk download - click topic to download Zip archive of all papers related to that topic: Acoustic Measurement Ambisonics Ambisonics, Evaluation of spatial audio, Multichannel and spatial audio, Psychoacoustics, Spatial audio applications Arrays/LSP Audio perception, Evaluation of spatial audio, Psychoacoustics, Sound design for immersive audio environments Audio Quality/Standards AV Art Education ELAC/Audio Tech Games/Interactive HRTF Music Analysis Psychoacoustics Psychology Reproduction, 3D Audio Reproduction: 3D Audio Synthesis
Film sound mixing is an iterative process where dubbing mixers combine several stems to form the final soundstage of a motion picture. For cinema: end to end control makes it possible to reproduce near consistent mixes at the playback stage, whereas in a domestic setting: playback hardware and listening levels vary greatly in addition to background sounds or noise in the listening space. While mixers have a general consensus on how their mixes translate in a domestic setting; there are a number of considerations to take into account when making mixing decisions at the post-production stage with little documented on best approaches for the task. This paper describes key considerations and requirements to take into account when mixing film sound to ensure translation in a domestic setting and a listening experiment for the purpose of testing different mixing approaches. Two different film mixing approaches were done.
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This paper describes an effective and fast solution for estimating speaker direction of arrival in 3D space. The main idea is to use noise as an excitation signal for sources and to determine time of arrivals from recordings of 3D space distributed microphones. Up-sampling and interpolation are applied to improve times of arrivals estimation, which are then used to construct sinusoidal signals. Sine phases are determined from times of arrival estimations and the frequency is optimised from the physical sensors’ structure. Hence, speaker direction are predicted by beamforming and direction of arrival algorithms with a precision of two degrees in azimuth and five in elevation.
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The sound of thunder is widely used in game, film and virtual reality sound design. It is also a phenomenon for which we seek a better understanding of the physics underlying the sound. Though many models of thunder have been proposed, there has not yet been a formal perceptual evaluation of the models to assess their realism and sound quality. Here, we present and evaluate the implementation of several thunder sound effect synthesis models. The models include different physical modeling and signal-based approaches, as well as a recorded sample. Evaluation was with over 50 participants. The results showed that none of the models were close to the recording in terms of realism, though signal-based models slightly outperformed the physical models. This highlights the need for comparative perceptual evaluation in sound synthesis, and identifies the limitations of current thunder simulation approaches.
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This e-Brief proposes a spot-microphone arrangement in the context of immersive audio productions for recording acoustic instruments. The presented technique enables reproduction of a single instrument with control of its auditory width in the horizontal and vertical dimension during post production. The geometrical arrangement as well as the required post processing for immersive loudspeaker-based audio productions is discussed. Additionally, a recording of ten individual instruments and two singers with three spot-microphone positions each was carried out in an anechoic chamber using the new method. The audio files and detailed documentation are provided under CC BY-NC-ND license for evaluation, demonstration and further research. An overview of this database as well as the initial findings gathered throughout mixing the 100+ track production for a 7+4 loudspeaker setup are given.
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Voice disorders due to strenuous usage of unhealthy voice qualities are a common problem in professional singing. In order to minimize the risk of these voice disorders, vital feedback can be given by making aware of one’s sung voice quality. This work presents the design task of a vowel and voice quality indication tool which can enable such a feedback. The tool is implemented in form of a VST plug-in. The plugin’s interface provides a graphical representation of voice quality and vowel intelligibility by means of two 2D voice maps. The voice maps allow a graphical distinction of three voice qualities (modal, breathy or creaky), and the representation of a sung vowel within the formant space spanned by the first and second formant frequency. The design process includes (i) building a ground truth dataset by using a modified speech synthesizer, (ii) linear prediction analysis, and (iii) the visualisation of the estimated vowel and voice quality by means of the 2D voice maps. The plugin’s code is available as open source to enable further development.
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With the rapid technological growth and availability of IOT devices, hearing loss can happen at an accelerated rate and becomes an increasing problem. The current hearing threshold curves, defined in the ISO 389-7 standard, do not take this into account and require a revision. This Engineering Brief will present the results of an ongoing hearing loss study using a novel remote hearing threshold detection method and evaluate its effectiveness. It is a modification of Bekesy's tracking method that is performed remotely on mobile devices to collect results on a large scale and simplify the measurement process. The results showed that the proposed method is effective for hearing loss detection and indicated that new hearing threshold curves can be defined. As the test is being actively distributed, by the time of publication it is expected to have sufficient subject size to represent the general population.
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During COVID, musical ensembles at high schools and colleges around the globe were forced to develop solutions to rehearsing while students were dispersed from campus or be forced to disband. The concept and required resources of telematic performance (performing with others online in real time) has been in development since the 1980s. The greatest challenge in telematic performance is reducing latency between musicians, enabling musicians to synchronize their performances with one another in real time. Recently, such platforms as JackTrip, Jamulus, and others have allowed musicians to perform together online with very low latency. But the possibility of performing online is dependent on having a stable and clean internet connection, plus the required audio equipment to capture a musician's performance. As was made evident during COVID, the digital divide must be negotiated and mediated as classrooms were migrated online. And musical performance additionally amplifies these challenges. This presentation will discuss and share the technological and unique social challenges of the migration of the York College Jazz Band to an online format.
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Conventional digital-to-analog conversion (DAC) is accomplished via a single processing path that must optimize broadband noise level against maximum output level, i.e., dynamic range. By splitting DAC processing into two or more discrete elements or “paths,” and passively recombining the analog resultants, order of magnitude improvement in dynamic range and linearity can be realized. Following a brief historical review, this paper will explore design details and experimental results on the author’s multi-path DAC prototypes, followed by experiments that assess multi-path design margins which exceed required psychoacoustic delivery parameters.
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