AES E-Library Search Results
[Feature] Electric guitar tone, you know it’s right when you hear it. How is it achieved? The typical starting approach at the guitar amp: Shure SM57 microphone, slightly off center of one of the cones of a driver, up close and almost touching the grille cloth. Oh, and angle the microphone a little. Ask veteran engineers why this microphone placement strategy is so common and a range of justifications follows, from seemingly scientific explanations, to vague guesses, to an honest, “I have no idea. I’ve always done it that way. Everyone does.”
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Perceptual Evaluation of Headphone Compensation in Binaural Synthesis Based on Non-Individual Recordings
The headphone transfer function (HpTF) is a major source of spectral coloration observable in binaural synthesis. Filters for frequency response compensation can be derived from measured HpTFs. Therefore, we developed a method for measuring HpTFs reliably at the blocked ear canal. Subsequently, we compared non-individual dynamic binaural simulations based on recordings from a head and torso simulator (HATS) directly to reality, assessing the effect of non-individual, generic, and individual headphone compensation in listening tests. Additionally, we tested improvements of the regularization scheme of an LMS inversion algorithm, the effect of minimum phase inverse filters, and the reproduction of low frequencies by a subwoofer. Results suggest that while using non-individual binaural recordings the HpTF of the individual used for the recordings – typically a HATS – should be used for headphone compensation.
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EspGrid is a protocol developed to streamline the sharing of timing, code, audio, and video in participatory electronic ensembles, such as laptop orchestras. An application implementing the protocol runs on every machine in the ensemble, and a series of “thin” helper objects connect the shared data to the diverse languages that live electronic musicians use during performance (Max, ChucK, SuperCollider, PD, etc.). The protocol/application has been developed and tested in the busy rehearsal and performance environment of McMaster University’s Cybernetic Orchestra, during the project “Scalable, Collective Traditions of Electronic Sound Performance” supported by Canada’s Social Sciences and Humanities Research Council (SSHRC), and the Arts Research Board of McMaster University.
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Perceptual Evaluation of Model- and Signal-Based Predictors of the Mixing Time in Binaural Room Impulse Responses
When creating virtual acoustic environments, the computational demands can be reduced by using generic late reverberation. Beyond the “mixing time,” the diffuse reverberation no longer contains details of the specific location. Therefore, a perceptually validated model for predicting the mixing time of different spaces will be helpful. This study evaluates various predictors of the perceptual mixing time using 9 different spaces. Both model- and signal-based estimators of mixing time were examined for their ability to predict the results of a group of expert listeners. For a shoebox-shaped room, the average perceptual mixing time can be predicted by the enclosure’s ratio of volume over surface area V/S and by vV, which serve as indicators of the mean free path length and the reflection density, respectively. Moreover, the “echo density profile” by Abel and Huang (AES paper 6985) can be used to predict the perceptual mixing time from measured data.
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Wave Field Synthesis (WFS) can synthesize virtual sound sources that are perceived to be at locations between loudspeakers and the listener, called focused sources. Because of practical limitations in the density of loudspeakers, there are artifacts. This research explores the amount of perceptual artifacts and the localization of the focused sources. The results from a variety of listening configurations illustrate the trade-offs. The truncation of loudspeaker arrays creates two opposite effects: (a) fewer additional wave fronts reduce the perception of artifacts, (b) stronger diffraction reduces the size of the listening area with adequate binaural cues.
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A new algorithm for 5.1 to stereo downmix is introduced that addresses the problem of dialogue intelligibility. The algorithm utilizes proposed signal processing algorithms to enhance the intelligibility of movie dialogue, especially in difficult listening conditions or in compromised speaker setup. To account for the latter, a playback configuration utilizing a portable device, i.e., an ultrabook, is examined. The experiments are presented that confirm the efficiency of the introduced method. Both objective measurements and subjective listening tests were conducted. The new downmix algorithm is compared to the output of a standard downmix matrix method. The results of subjective tests prove that an improved dialogue intelligibility is achieved.
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In this paper we discuss how software development can be improved in the audio and music research community by implementing tighter and more effective development feedback loops. We suggest first that researchers in an academic environment can benefit from the straightforward application of peer code review, even for ad-hoc research software; and second, that researchers should adopt automated software unit testing from the start of research projects. We discuss and illustrate how to adopt both code reviews and unit testing in a research environment. Finally, we observe that the use of a software version control system provides support for the foundations of both code reviews and automated unit tests. We therefore also propose that researchers should use version control with all their projects from the earliest stage.
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Sound atmospheric attenuation is a relevant aspect of realistic space modeling in 3-D audio simulation systems. A digital filter has been developed on commercial DSP processors to match air absorption curves. This paper focuses on the algorithm implementation of a digital filter with continuous roll-off control, to simulate high frequency damping of audio signals in various atmospheric conditions, along with rules to allow a precise approximation of the behavior described by analytical formulas.
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Polar Measurements of Harmonic and Multitone Distortion of Direct Radiating and Horn Loaded Transducers
While extensive literature is available on the topic of polar pattern measurements and predictions of loudspeakers’ fundamental SPL, only a single paper to our knowledge deals with the polar pattern of nonlinear distortions, in particular with harmonic distortion products of cone type loudspeakers. This paper contains the first results of a more thorough study intended as a complement to fill the gap both in measurement techniques and loudspeaker type. Relative and absolute harmonic distortion as well as relative and absolute multitone distortion, indeed, have been measured for cone, dome, and horn loaded transducers.
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A time-frequency-domain non-linear parametric method for spatial audio processing is presented here, which can utilize microphone input having directional patterns of any order. The method is based on dividing the sound field into overlapping or non-overlapping sectors. Local pressure and velocity signals are measured within each sector, and an individual Directional Audio Coding (DirAC) processing is performed for each sector. It is shown, that in certain acoustically complex conditions the sector-based processing enhances the quality compared to traditional first-order DirAC processing.
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