Bulk download - click topic to download Zip archive of all papers related to that topic: No Subject Listed Acoustics Applications in Audio Audio Education Cinema Sound Forensic Audio Game Audio Listening, Hearing, & Production Perception Recording and Production Room Acoustics Signal Processing Sound Reinforcement Spatial Audio Transducers Transducers/Perception
In normal conditions (e.g., a living room) the total sound power emitted by the loudspeaker plays an important role in the listening experience. Along with the direct sound and first reflections, the sound power defines the loudspeaker performance in the room. The acoustic resonances of the loudspeaker system are especially important, and thanks to spatial averaging, are more easily revealed in the sound power response. In this paper we use time-frequency analysis to study the spatially averaged impulse response and reveal the structure of its resonances. We also show that the net effect of loudspeaker equalization is not only the attenuation of the resonances but also the shortening of their duration.
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Recently the ISO/IEC MPEG-H 3D Audio standard for perceptual coding of one or more audio channels has been finalized. It is a little-known fact that, particularly for communication applications, the 3D Audio core-codec can be operated in a low-latency configuration in order to reduce the algorithmic coding/decoding delay to 44, 33, 24, or 18 ms at a sampling rate of 48 kHz. This paper introduces the essential coding tools required for high-quality low-delay coding–transform splitting, intelligent gap filling, and stereo filling–and demonstrates by means of blind listening tests that the achievable subjective performance compares favorably with, e.g., that of HE-AAC even at low bit-rates.
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Dialog is often considered the most important audio element in a movie or television program. The potential for artifact-free dialog salience personalization is one of the advantages of new object-based multichannel digital audio formats, along with the ability to ensure that dialog remains comfortably audible in the presence of concurrent sound effects or music. In this paper we review some of the challenges and requirements of dialog control and enhancement methods in consumer audio systems, and their implications in the specification of object-based digital audio formats. We propose a solution incorporating audio object loudness metadata, including a simple and intuitive consumer personalization interface and a practical head-end encoder extension.
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Narrow band parametric speech coding and wideband audio coding represent opposite coding paradigms involving audible information, namely in terms of the specificity of the audio material, target bit rates, audio quality, and application scenarios. In this paper we explore a new avenue addressing parametric coding of wideband speech using the potential and accuracy provided by frequency-domain signal analysis and modeling techniques that typically belong to the realm of high-quality audio coding. A first analysis-synthesis validation framework is described that illustrates the decomposition, parametric representation, and synthesis of perceptually and linguistically relevant speech components while preserving naturalness and speaker specific information.
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Single-band shelving or presence boost/cut filters are useful building blocks for a wide range of audio signal processing functions. Digital filter coefficient formulas for elementary first- or second-order IIR parametric equalizers are reviewed and discussed. A simple modification of the classic Regalia-Mitra design yields efficient solutions for tunable digital equalizers whose dB magnitude frequency response is proportional to the value of their gain control parameter. Practical applications to the design of tone correctors, artificial reverberators and environmental audio signal processors are described.
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The diffusion equation model has been intensively researched as a room-acoustics simulation algorithm during last years. A 3-D finite-difference implementation of this model was proposed to evaluate the propagation over time of sound field within rooms. Despite the computational saving of this model to calculate the room energy impulse response, elapsed times are still long when high spatial resolutions and/or simulations in several frequency bands are needed. In this work several data-parallel approaches of this finite-difference solution on Graphics Processing Units are proposed using a compute unified device architecture programming model. A comparison of their performance running on different models of Nvidia GPUs is carried out. In general, 2D vertical block approach running in a Tesla K20C shows the best speed-up of more than 15 times versus CPU version.
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We derive a novel explicit wave-domain model for “diode clipper" circuits with an arbitrary number of diodes in each orientation, applicable, e.g., to wave digital filter emulation of guitar distortion pedals. Improving upon and generalizing the model of Paiva et al. (2012), which approximates reverse-biased diodes as open circuits, we derive a model with an approximated correction term using two Lambert W functions. We study the energetic properties of each model and clarify aspects of the original derivation. We demonstrate the model's validity by comparing a modded Tube Screamer clipping stage emulation to SPICE simulation.
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Most electrical engineering (EE) students are familiar with LabVIEW. This graphical programming environment is commonly used in university EE educational and research labs to facilitate data acquisition and processing using a suite of built-in mathematical, DSP, and communication functions. LabVIEW is particularly adept at emulating control panels with a variety of knobs, sliders, and gages. The audio functionality of LabVIEW, along with its “knobby” user interface, makes it ideal for exploration of music synthesis concepts by EE students. In this paper several types of music synthesis are explored in LabVIEW. Implementation of these in elective EE coursework gives EE students a unique opportunity to experience abstract concepts such as waveforms, frequency, filtering, and envelopes through their auditory cortex, reinforcing what they have learned through traditional pedagogy, and also provides EE students an introduction to some basic audio engineering (AE) concepts.
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Traditional barriers of geography, organization, and culture and being broken down by emerging technology [1]. In the recording industry, professionals often collaborate on projects globally, engaging in what Tapscott and Williams [2] call “peer-production.” The potential in these concepts extends to those developing their expertise—they can connect with peers and industry professionals on a global scale. Despite the potential however, most Higher Education institutions fail to engage for cultural reasons. This paper outlines a model for collaborative learning explored and developed through a project funded by the UK's Higher Education Academy. The project involved Coventry University and industry organization JAMES as well as a number of other international partners. The paper looks at the pedagogical background to the project, some typical activities undertaken before summarizing the key outcomes and opportunities for further work.
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Many students enrolling in audio programs are not fully aware of the importance of science for the audio professional. Typically these students are creative but may have deficiencies in math and science. My goal as an instructor is to minimize the negative associations of these subjects through active lesson plans that stress practical audio situations in a compelling and interactive manner. As a result, students develop confidence through their ability to use science as a tool to both solve audio problems and create expressive art forms. My approaches empower them to succeed in early courses, which facilitate creative applications in later classes. Consequently, students are better prepared for the job force using skills that promote both technical and creative capacities.
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