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Bulk download - click topic to download Zip archive of all papers related to that topic:   3D/Immersive/Spatial Audio    Audio Synthesis & Audio Effects    Binaural Audio    Extended Reality Audio    Loudness & Perception    Loudspeakers and headphones    Machine Learning / Artificial Intelligence    Network Audio    Recording Technologies    Room Acoustics    Sound Classification    Sound Quality & Perception    Spatial Audio    Studio Technology    Television Audio   

 

Latency and Quality-of-Experience Analysis of a Networked Music Performance Framework for Realistic Interaction

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Networked Music Performances (NMPs) are of increasing importance for creative and cultural professionals exploring new solutions and opportunities to perform at geographically distant locations. Although latency is one of the most important parameters for the transmission of audio information, it is rarely possible to obtain insights into the individual components of the transmission latency during NMP. In this publication a detailed evaluation of the latency budget of an ultra-low latency audio transmission in an NMP system for realistic interaction, build up between Hanover and Munich (˜500 km), is presented. To explore the Quality-of-Experience (QoE) of the NMP, objective results of a pop / rock piece played by five professional musicians are compared with their subjective perception. Measures of network performance, as well as the comparison of the objective musical results with the subjective user feedback suggest that the musicians had an experience close to a real performance.

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“We’ll Feel That in Post”: Altering Emotional Perception With Different Piano Presentations In the Context of Lyrics

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While much is known about how musical performers can communicate emotions through music, less research has been dedicated to determining if audio processing can affect the emotion of a musical work as it is perceived by the listener, particularly within the context of a narrative provided by sung lyrics. This paper presents a pilot experiment in which 8 participants were presented with two audio recordings of the same piano and vocal performance, with the piano presented with different timbral characteristics in each recording. Results demonstrate some limited correlation between change in timbre of the piano and change in perceived emotion by the listener.

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Reduction of 3D Ambisonic to 2D using plane-wave decomposition

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This articles deals with the reduction – or “projection” or “downscaling” – of a 3D-encoded Ambisonic sound field (“full-sphere” or “periphonic”) into a 2D representation (“horizontal-only” or “planar” or “pantophonic”). We show that the reduction operation can be equivalently achieved by 1) applying conversion formula of the normalization factors, or 2) performing a plane-wave decomposition of the original sound field and re-encoding the resulting plane waves to 2D Ambisonic. The latter approach provides greater flexibility in adapting the content to horizontal-only reproduction.

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Spectral and spatial perceptions of comb-filtering for sound reinforcement applications.

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Most sound reinforcement systems consist of multiple loudspeakers systems arranged strategically to cover the entire audience area. This study investigates the spectral and spatial perceptions of interferences that can be experienced in the shared coverage area between two full-range loudspeakers. A listening test was conducted to determine the effect of lag source delay, relative level, and angular separation, on the perception of spectral coloration and spatial impressions (width, localization shift, image separation). The results show that spectral coloration is considerably reduced when sources are spatially separated, even with a small azimuth angle (10°). It was also found that coloration audibility depends on the interaction between the audio track and the delay introduced. Finally, the type of perceived spatial degradation depends mainly on the spatial separation and on the relative level of the source arriving later in time (lag source).

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Ambisonic Decoder Test Methodologies based on Loudspeaker Reproduction

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The comparative evaluation of the quality of different Ambisonic decoding strategies presents a number of challenges, most notably the lack of a suitable reference signal other than the original, real-world audio scene. In a previous paper, a new test methodology for such evaluations was presented via a listening test conducted using binaural reproduction. In this paper, this methodology is further refined and the results of a new listening test using loudspeaker reproduction over a 7.0.4 array is presented. The results again indicate some significant differences between the decoders for certain attributes.

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Twang! A physically derived synthesis model for the sound of a vibrating bar

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A physically derived synthesis model of the sound generated when a ruler is twanged while hanging over the edge of a solid surface is presented. This is a sound effect used in movies, TV, theatre performances and cartoons. The model is derived from the Euler-Bernoulli equation, offering the user a set of physical parameters to control ruler length as well as the material properties. Perceptual evaluation indicates that the model can be perceived as realistic as a recorded ruler twang as well as being able to replicate sounds of similar quality as an alternative synthesis model.

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Velocity-Contolled Parameter Switching for Echo Cancellation in Immersive Telepresence with Continuously Changing Microphone Positions

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The frequency-domain adaptive Kalman filter (FDAKF) is a popular choice for multichannel acoustic echo cancellation (AEC) due to its good initial convergence and robustness to double-talk. However, without additional measures its reconvergence and tracking capabilities are known to be suboptimal. Previous studies have particularly focused on abrupt echo path changes and have proposed different methods to optimize the filter’s reconvergence. Motivated by our application of an acoustic echo cancellation system for immersive telepresence, this paper investigates continuous echo path changes caused by moving microphones. The echo cancellation performance of the FDAKF is studied for different parameters of the underlying model inside the Kalman filter. Experimental results show, that even in the very challenging scenario of a moving microphone, a small echo reduction can still be achieved with suitable parameters for the considered microphone velocities. Furthermore, a novel method is proposed, which includes a microphone motion-controlled online parameter switching for the FDAKF by means of external motion sensors. In this paper the method is studied within a proof-of-concept. Experiments show a behavior matched to static and dynamic phases and even an increased reconvergence speed in the transition from dynamic to more static phases.

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MMAD – Designing for Height – Practical Configurations

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Although the basic philosophy behind the design of Microphone arrays, for 3D audio recording and reproduction, has been described in previous AES papers[1][2][3][4], no specific examples have been given with respect to various Surround Sound arrays and the corresponding height arrays (1st layer of height array microphones and the Zenith microphone). This paper gives four examples of complete 3D Audio arrays with perfect critical linking. Examples include same microphone directivity arrays, as well as hybrid arrays. Suitable steering functions are discussed, and specific values are given for each array combination, so as to obtain perfect critical linking.

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Survey of User Perspectives on Headphone Technology

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Headphones are widely used to consume media content at home and on the move. Developments in signal processing technology and object-based audio media formats have raised new opportunities to improve the user experience by tailoring the audio rendering depending on the characteristics of the listener’s environment. However, little is known about what consumers consider to be the deficiencies in current headphone-based listening, and therefore how best to target new developments in headphone technology. More than 400 respondents worldwide took part in a headphone listening experience survey. They were asked about how headphones could be improved, considering various contexts (home, outside, and public transport) and content (music, spoken word, radio drama/tv/film/online content, and telecommunication). The responses were coded into themes covering technologies (e.g. noise cancellation and transparency) and features (e.g. 3D audio) that they would like to see in future headphones. These observations highlight that users’ requirements differ depending on the listening environment, but also highlight that the majority are satisfied by their headphone listening experience at home. The type of programme material also caused differences in the users’ requirements, indicating that there is most scope for improving users’ headphone listening experience for music. The survey also presented evidence of users’ desire for newer technologies and features including 3D audio and sharing of multiple audio streams.

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Controlling the Balance Between Early and Late Reflections of an Impulse Response Using the Modal Decomposition Paradigm

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Previous work has outlined a method of decomposing an acoustic impulse response (IR) into decaying sine components, which allows for parametrization of a modelled IR. This enables control over a convolution based reverberator akin to an algorithmic reverberator. A common control found in an algorithmic reverb is the balance between early and late reflections. This work introduces a method of adjusting the balance between early and late reflections without direct processing of the impulse response, through a component domain transform. This work extends the creative applications of indirect IR Processing through the use of the modal decomposition paradigm.

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                 Search Results (Displaying 1-10 of 69 matches)
AES - Audio Engineering Society