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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   

 

Non-Ideal Operational Amplifier Emulation in Digital Model of Analog Distortion Effect Pedal

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Digital models of analog guitar effects pedals have largely ignored the impact of non-ideal components on the resulting timbre, though the physical limitations of analog components are sometimes key to achieving the intended effect. The signature sound of the Pro Co RAT is largely attributed to the non-ideal characteristics of the Motorola LM308 operational amplifier, particularly the slew-rate, gain-bandwidth product and supply voltage. Analysis of harmonic and spectral content shows that the inclusion of these non-ideal component characteristics results in a more accurate recreation of the Pro Co RAT distortion effect. In a comparison of real-time digital models, the additional computational cost of the non-ideal model was negligible.

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A dataset of measured spatial room impulse responses in different rooms including visualization

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In this contribution, an open-source dataset of captured spatial room impulse responses (SRIRs) is presented. The data was collected in different enclosed spaces at the Technische Universität Ilmenau using an open self-build microphone array design following the spatial decomposition method (SDM) guidelines. The included rooms were selected based on their distinctive acoustical properties resulting from their general build and furnishing as required by their utility. Three different classes of spaces can be distinguished, including seminar rooms, offices, and classrooms. For each considered space different source-receiver positions were recorded, including 360? images for each condition. The dataset can be utilized for various augmented or virtual reality applications, using either a loudspeaker or headphone-based reproduction alongside the appropriate head-related transfer function sets. The complete database, including the measured impulse responses as well as the corresponding images, is publicly available.

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Developing a Binaural Renderer for Audio Definition Model Content

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The Audio Definition Model (ADM) can be used to represent object- and scene-based audio programmes, and there is a standardised method for reproducing ADM content on loudspeakers, but not currently for headphones. We present the design and implementation of a binaural renderer for the ADM, which supports the features of the ADM while maintaining high-quality output. For rendering objects the system uses virtual loudspeaker rendering with windowed binaural room impulse responses (BRIRs). To reduce comb-filtering effects, delay is removed from the BRIRs and replaced with a per-ear and per-object variable fractional delay line. When rendering diffuse sources, the original delays are used, as the varied onset delays help create perceived extent. The overall gain of each source is adjusted dynamically to compensate for loudness changes caused by interactions between BRIRs of neighbouring loudspeakers. The system is available as an open-source C++ library based on the VISR framework, suitable for adding real-time head-tracked binaural output to applications, and is built into the EAR Production Suite.

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Design of a lightweight acoustical measurement room

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The paper presents the design principles of an acoustic test chamber, where the insulation requirements of typical measurement rooms are relaxed and so constructing the surfaces using very lightweight materials, consisting only of absorbents and a simple frame, is possible. The test chamber constructed according to these principles shows good absorption characteristics down to 200Hz and has a significantly larger free space for measurements than a conventional chamber designed using wedges and solid walls.

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Designing sound system in-band headroom based on expected difference between C- and A-weighted levels

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Sound pressure level (SPL) is the standard metric for regulations regarding environmental noise exposure. Because performances are often regulated by their A-weighted sound level, it is tempting to think that A-weighted level should be the primary design consideration for sound system headroom. Because A-weighting disregards significant low-frequency energy, it is possible to create a wide variety of spectra with the same A-weighted level, but each having a different spectral shape and C-weighted level. While regulators correlate excessive A-weighted levels with hearing damage, A-weighted levels are less well correlated with community annoyance. The Netherlands has recognized this and created a permitting system incorporating the difference between C- and A-weighted sound levels (C-A) as a measure of low-frequency content. This Brief gives supporting evidence for the correlation between C-A levels and different musical genres and offers complementary design guidance corresponding to sound system headroom with emphasis on in-band levels.

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Overview of Evaluation Methods of Sound Field Reproduction Systems

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Sound reproduction is a task where the goal is to accurately reproduce sound field of a previously captured audio scene in a defined controlled receiving space. According to the current state-of-the-art technology there is no system available which can perform such task perfectly. Unrestricted reproduction of a three-dimensional sound field would require a spatially continuous sound source, therefore any real sound field reproduction system, consisting of discrete sound sources, offers only an approximation of the original sound field. Each system can be objectively evaluated against various aspects of sound field reproduction accuracy like spectral and level matching, quality and distortions, time, and directivity cues, "sweet spot" size, the influence of obstacles in the receiving space (like equipment or people) or using perceptual objective metrics. The goal of this work is to compile these evaluation methods and provide mapping to the abovementioned key aspects of sound field reproduction. The authors contextualize the overview in the acoustic testing perspective, where specific aspects of a sound reproduction accuracy matter in evaluation of different audio-related feature of a device (e.g., for testing quality of Direction of Arrival – directivity ques are critical for evaluation, whereas for testing Acoustical Scene Classification – spectral and level accuracy is more relevant) but the evaluation methods and findings can be translated into other areas of audio research.

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Virtual Studio Production Tools with Personalized Head Related Transfer Functions for Mixing and Monitoring Dolby Atmos and Multichannel Sound

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With the increasing popularity of audiophile headphones in this decade, the need for mixing over headphones is on the rise. Studio engineers use headphones as a critical tool for checking their mixes over the headphones before publishing them. As dolby atmos music and surround sound music is currently regaining popularity, there is also an increasing need for having multi channel speaker setups and associated gear in the studio to produce music in such formats. Such systems are extremely expensive and time consuming to set up. In this engineering brief, we present virtual studio production tools for mixing and monitoring atmos and multichannel sound with personalized head-related transfer functions (HRTFs). This paper talks in detail how the acoustics of the studio, including speaker, and headphone responses are captured accurately for a truly immersive experience. The acoustic fingerprint of the studio is then integrated with the personalized HRTFs predicted using machine learning algorithms that use an ear image as an input. Such novel tools will bring the power of personalized spatial audio and dolby atmos production in hands of millions of at-home mixing engineers and producers.

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Applause detection filter design for remote live-viewing with adaptive modeling filter

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The COVID-19 pandemic prevents us from enjoying live performances. On the other hand, commercial audio-visual transmission systems, such as live viewing systems, have become more popular and have been increasing. The APRICOT: (APplause for Realistic Immersive Contents Transmission) system was developed and used in some trials to enhance the reality for live viewing. This paper describes an applause sound extraction method for automation of applause sound transmission and a simulation experiment using the sound source recorded live at the venue to assess the applause sound extraction performance. We used an adaptive filter to model the room transfer function. In addition, we designed the inverse filter to emphasize applause sounds and extracted them. The experimental evaluation showed that the system extracted the applause sounds almost correctly under various conditions from the performance sound source.

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Numerical and Experimental Analysis of a Metamaterial-based Acoustic Superlens

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For many years, the engineering limitations in a single loudspeaker have offered no solution to the problem of delivering sound only to parts of an audience. Precise control on how sound is delivered to an audience has required multiple loudspeakers, either through their distribution or through DSP. The recent uptake of acoustic metamaterials, however, seem to offer different solutions. Using devices based on acoustic metamaterials, for instance, brings to acoustics design principles that come directly from optics, at a reasonable manufacturing cost. In this work, we design, numerically simulate, and characterise an acoustic converging superlens: a 3D-printed device capable of focusing an incoming plane wave at a distance less than one wavelength. We show how a loudspeaker at a fixed distance from the lens results in an “image” of the source at a distance prescribed by the thin-lens equation. Finally, we propose possible applications of such an acoustic superlens to future audio experiences.

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Phase Mitigation Through Filter Design

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In both acoustic and digital systems, delays and the resulting phase interference are an innate feature of sound recording; traditionally, phase-interference mitigation is applied through temporal offset to attempt time coherence between multiple signal paths. Filter design presents an alternative solution to phase issues, wherein predictive modeling allows for a filter to apply corrective magnitude response. Such application of filter design presents its own set of problems and could further be explored in creative, rather than remedial, settings.

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