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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   

 

Dynamic Range Controller Ear Training: Transfer of Learned Skills to a Related Task

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A technical ear training program designed to teach participants to identify the audible effects of dynamic range compressors was developed and used to train graduate students in music production. A pre-/post-training listening test was used to determine if the students could transfer skills learned during the training program to a related listening task. The participants executed the task repeatedly and the participants’ variance in their final compressor settings was measured. The pre-/post-tests were administered to the trained student group, an untrained student control group, and a group of recently graduated professional engineers. A reduction in variance between the pre-/post-test was measured in the trained group but not in the control group. The trained students also had less variance in their responses than the professional engineers.

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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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Geometrical Acoustics Approach to Cross Talk Cancellation

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Crosstalk Cancellation (CTC) is a signal processing technique allowing for immersive sound reproduction from a limited number of loudspeakers. Pioneered in the sixties, CTC has lately gained much attraction due to upcoming Augmented Reality and Virtual Reality applications and generalization of 3D audio content. In this paper, we present a novel time-domain approach to CTC based on modeling of the system’s geometrical acoustics. Our solution provides a simple processing model, as well as means to address robustness issues and adaptation to arbitrary listener positions.

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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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Multi-diaphragm Micro-speaker vs. Single-diaphragm Micro-speaker

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With wearable devices becoming increasingly popular, there is a foreseeable paradigm shift for small format or miniature audio transducers in the near future. Small single diaphragm micro-speaker transducer no longer meets the new expectations both in low frequency performance and overall sound pressure level. In this paper a new multi-diaphragm micro-speaker is presented. The acoustic performance of the transducer is simulated using finite element method. Furthermore, this paper analyses the low frequency performance between the single diaphragm micro-speaker and the multi-diaphragm micro-speaker.

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