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Bulk download - click topic to download Zip archive of all papers related to that topic:   Audio Applications and Technologies    Audio Content Management    Audio Equipment    Audio Processing    Immersive Audio    Perception    Recording, Production, and Reproduction   

 

An approach for implementing time-stretching as a live realtime audio effect

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This paper presents a novel buffer management technique that allows time-stretching to be implemented as a live realtime audio effect. Previously, implementing time-stretching as an effect in a realtime audio system with live input and output was not practical due to technical and perceptual problems. This system is based on the circular buffer, but with a non-continuous, audio level triggered write methodology. This modification compensates for the fact that the output of a time-stretcher is longer than the input. This allows synchrony between a live input and output to be maintained as well as reducing the processing and memory requirements. The buffer management technique is paired here with the Paulstretch1 time-stretch algorithm, but can be adapted for use with other time-stretch algorithms as well. Also discussed are the implementation design details including live, manipulatable FFT window size and stretch factor parameters.

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Derivation of horn parameters from numerical simulations, analytical methods, and measurements

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In the authors’ previous work, the input acoustical impedance of an axisymmetric horn was obtained through the combination of matrix analysis and dedicated electro-acoustical measurements. This work extends the approach to further refine the derivation of the acoustical impedance and transfer function of several different horns. The results are evaluated against COMSOL modeling and analysis made through the two-microphone method. A one-dimensional matrix model is also used for the horn impedance derivation. Acoustical transfer functions of two different horns are measured and compared with COMSOL modeling results.

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Profiling loudspeaker nonlinearities using parametric multitone stimulus

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The previous work reported on the use of modulated musical multitone (MMM) for the detection of audible distortion in horn drivers. With an adjustable crest factor, MMM is designed to match the properties of music and is used to measure nonlinear distortion audibility thresholds. In this work, the signal is used to create a distortion profile of two-way systems for comparison. The MMM reaction is post-processed to obtain the Multitone Total Nonlinear Distortion response, which presents nonlinear distortion products averaged in a “sliding window” in the frequency domain. MTND responses are derived from a low level to a very high level. These MTND curves are compiled into a comprehensive spectral distortion profile which can be used to compare multiple systems.

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Design and Evaluation of a Binaural Chorus Effect

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A binaural chorus effect is designed in Max/MSP and Python, aiming to take the chorus effect beyond its current use in stereo mixes. Six separate iterations of the design are evaluated alongside a stereo version, focusing on how changes in azimuth position and at what point binaural synthesis is rendered in the signal processing affects the final product. Results are not found to be conclusive enough to set design standards, but are conclusive enough to provide a baseline methodology for future works.

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Open Database of Spatial Room Impulse Responses at Detmold University of Music

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This manuscript presents an open source database of Spatial Room Impulse Responses (SRIR) captured at three different performance spaces of the Detmold University of Music. It includes one medium sized concert hall (Detmold Konzerthaus), one chamber music room (Brahmssaal) and one theater (Detmold Sommertheater). The collection contains approximately 600 multichannel RIRs corresponding to several source and receiver configurations. For each room we include measurement positions on stage and at the audience area captured with both an artificial head and an open microphone array compatible with the Spatial Decomposition Method (SDM). The Detmold Konzerthaus holds a large scale Wave Field Synthesis system, and SRIRs of an ensemble of focused sources on stage and conditions of increased reverberation are also included.

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ArcheoEchi – a virtual reconstruction of a medieval cathedral in Southern Italy

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ArcheoEchi is an interactive virtual reality experience for Oculus Go based on the medieval cathedral of Saint Albert, located in the Southern Italian archeological site of Montecorvino (Foggia). The project saw the collaboration of the Department of Humanities of the University of Foggia and the AudioLab of the University of York, and it concerned the 3D modeling of the church, the auralization of its acoustics and the implementation of a binaural listening system. The development contributed to raise new questions about architectural acoustics of ancient heritage and implement new solutions for designing interactive systems. Thanks to both its educational value and academic background, ArcheoEchi shows how immersive technologies can enhance the offer of cultural heritage and inspire new academic studies.

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Design and Application of a Native-D Recording Format for Optimal Dolby Atmos Reproduction

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In this paper I will propose a Native-D format immersive microphone system made specifically for mixing and reproduction within the Dolby Atmos framework. While there are a number of immersive recording systems in use today, most have been designed for reproduction within listening environments that use sub 90° angled height channels such as Auro-3D. The microphone system proposed in this paper aims to take advantage of the native 90° height-to-main layer angle found in Atmos speaker systems by combining a non-coincident main layer with a corresponding near-coincident, 90° directional microphone for each height channel. This system excels in its high-fidelity capture of both group and individual sources, and benefits from a high level of decorrelation from channel to channel. An additional advantage of the system is its capture of Native-D format signal, which ensures that no format conversion or complex matrixing must be done, avoiding a loss of fidelity from recording to mixing stages. A case study of this system, consisting of multiple recording sessions, has been done in order to establish the validity of a system of this type, ultimately resulting in a final mix using Dolby Atmos. Both a “native immersive” and a “non-native” recording approach were taken in order to exemplify the system’s versatility.

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Investigation of a Real-Time Hearing Loss Simulation for use in Audio Production

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We present a perceptually motivated, real-time hearing loss simulation for use in audio production. The implementation builds on a previous simulation, but is now real-time, low latency, and available as a stereo audio effect plug-in with more accurate modelling of hearing loss. It offers the option of isolating and customizing high frequency threshold attenuations on each ear, corresponding to the audiogram information. The simulation also provides the option of incorporating additional suprathreshold effects such as spectral smearing, rapid loudness growth and loss of temporal resolution on audio. The underlying psychoacoustic principles are described, and results are presented to show the simulation’s performance.

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Loudspeaker characterization by indirect measurement of diaphragm velocity

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A complete characterization of an electrodynamic transducers consists, among others, of measurement of the membrane velocity. Modern measurement devices, such as laser vibrometers and accelerometers, are coming with their set of disadvantages. The laser devices can be costly, the accelerometers, being attached to the device under test, are influencing the measurement and are unusable for small-sized loudspeakers. This paper explores the indirect measurement of the membrane velocity, using a single microphone, and the limits for measurement of loudspeaker parameters, including the frequency-dependent ones due to creep effects and eddy currents.

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Designing a 9-channel location sound microphone from scratch

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The design of a 9-channel microphone system for location recording of mainly atmospheres will be described. The key concept is matching the recording and reproduction angles of the individual sectors. The rig is designed for the AURO-3D 9-channel playback system (4 height speakers). An analysis of the reproduction angles will be included, as well as recording concepts like the Stereo Recording Angle (SRA), Williams curves, Scale Factors for different reproduction angles than 60° and diffuse field decorrelation. Finally, practical aspects like microphone mounts and windshields for such a system will be presented.

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                 Search Results (Displaying 1-10 of 25 matches)
AES - Audio Engineering Society