Authors:Straube, Florian; Schultz, Frank; Bonillo, David Albanés; Weinzierl, Stefan
Affiliation:Audio Communication Group, Berlin, Germany
Line source arrays (LSAs) are used for large-scale sound reinforcement to synthesize a homogeneous field over the full audio bandwidth. Sound reinforcement in different venues requires adapted curving of LSAs. A purely analytical approach for finding appropriate LSA cabinet tilt angles is presented in this article. Polygonal audience line curving (PALC) is based on the geometry of the receiver area and the intended coverage. In comparison with typical standard LSA curving schemes, PALC appears to be superior due to its flexible adaptability with respect to the receiver geometry. The deployed loudspeaker cabinets are rigged with different tilt angles and/or electronically controlled. This provides the intended coverage of the audience zones and avoids radiation toward the ceiling, reflective walls, or residential areas. PALC can be applied in advance of a numerical optimization of the loudspeakers’ driving functions. The method can be used with different objectives, such as a constant interaction between adjacent cabinets with respect to the receiver geometry. The advantages of the presented approach regarding sound field homogeneity and target-oriented radiation are evaluated based on technical quality measures.
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Authors:Hashemgeloogerdi, Sahar; Bocko, Mark F.
Affiliation:University of Rochester, Rochester, NY, USA
A room impulse response (RIR) characterizes the sound propagation between a source and a microphone placed in a room. Accurate modeling of an RIR is essential in many acoustic signal processing applications, and precise modeling of a RIR using a small number of parameters is often required. Modeling is a challenge because room responses may have tens of thousands of taps that vary greatly when the source and microphone locations are slightly changed. In this paper, a subband multichannel method for accurate modeling of long RIRs is proposed, which is computationally efficient and robust against RIR variations. A dual-tree complex wavelet packet transform is utilized to decompose a multichannel RIR into aliasing-free subband signals. Low-order adaptive Kautz filters are designed to model subband signals using the acoustical poles common to the RIR channels. A least-squares algorithm is introduced to efficiently estimate the common acoustic poles (CP) in each subband. The algorithm precisely estimates the CPs after a low number of iterations, and unconditionally guarantees the stability of the estimated poles. Experimental results demonstrate that the proposed method accurately models the room responses while exhibiting robustness against room response variations caused by changing the source and microphone locations.
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Authors:Dahl, Nicolai; Iversen, Niels; Knott, Arnold; Andersen, Michael A. E.
Affiliation:Technical University of Denmark, Kgs. Lyngby, Denmark
During the last decade, switch-mode audio amplifiers have become a common choice for audio applications because of efficiencies approaching 90% and distortions as low as 0.001%. Such amplifiers modulate the input audio into a high-frequency discrete signal that drives a Class-D power stage. The control loop is the key element in achieving high-quality performance. Modern control theory methods were used to design and simulate a full-state feedback integrating controller for use with a high-frequency bridge class-D amplifier. An optimal linear full-state integral controller based on the state-space model was designed using the Linear Quadratic Regulator (LQR) method, and verified on a linear and switching model. Measurements on a Class-D amplifier with the implemented controller showed that the step responses and THD+N measurements were aligned with theoretic predictions. A 30-fold reduction in THD+N was observed compared to open-loop. The results prove that the principals of modern control achieve good performance in Class-D amplifiers, even when the output filter has a large resonance.
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Authors:Woodcock, James; Davies, William J.; Melchior, Frank; Cox, Trevor J.
Affiliation:University of Salford, Salford, United Kingdom; BBC R&D, Dock House, MediaCityUK, Salford, United Kingdom
Object-based audio (OBA) is an approach to sound storage, transmission, and reproduction whereby individual audio objects contain associated metadata information that is rendered at the client side of the broadcast chain. For example, metadata may indicate the object’s position and the level or language of a dialogue track. An experiment was conducted to investigate how content creators perceive changes in perceptual attributes when the same content is rendered to different systems and how they would change the mix if they had control of it. The main aims of this experiment were to identify a small number of the most common mix processes used by sound designers when mixing object-based content to loudspeaker systems with different numbers of channels and to understand how the perceptual attributes of OBA content changes when it is rendered to different systems. The goal is to minimize perceived changes in the context of standard Vector Base Amplitude Panning and matrix-based downmixes. Text mining and clustering of the content creators’ responses revealed 6 general mix processes: the spatial spread of individual objects, EQ and processing, reverberation, position, bass, and level. Logistic regression models show the relationships between the mix processes, perceived changes in perceptual attributes, and the rendering method/speaker layout. The relative frequency of different mix processes was found to differ among categories of audio object, suggesting that any downmix rules should be object category specific. These results give insight into how OBA can be used to improve listener experience.
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Authors:Schinkel-Bielefeld, Nadja; Zhang, Jiandong; Qin, Yili; Leschanowsky, Anna Katharina; Fu, Shanshan
Affiliation:Fraunhofer Institute for Integrated Circuits IIS, Erlangen, Germany; Academy of Broadcasting Planning, SAPPRFT, Beijing, P.R. China; Pleasant Audio Consulting Ltd., Beijing, P.R. China
Using a standard protocol and sample audio cases to enhance reproducibility, tests of coding quality are often performed jointly by laboratories around the world. Multiple Stimuli with Hidden Reference and Anchor (MUSHRA) is one such standard protocol. The same audio samples are used in all labs and as a result, listeners inevitably are judging quality in either their native language or one that they do not understand. It is not clear if a lack of understanding the language and its phonemes can influence the listener’s perception and his or her quality ratings during the test. This study used German and Mandarin Chinese speaking listeners, as well as test material in these two languages. The authors analyzed how ratings and listening times were affected by the foreign language. When results were pooled over all conditions, no significant differences between the ratings were found. However, for items of high audio quality, it was observed that listeners needed more time to evaluate samples that were not their native language, and it took more effort to compare different audio signals. As in MUSHRA tests – contrary to ITU-T P.800 tests - listeners can compensate for any difficulty they may have in perceiving artifacts by more effort and longer listening times, it seems to be no problem to include nonnative listeners in these tests at the expense of making them slightly less efficient.
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Authors:Nease, Stephen H.; Lanterman, Aaron D.; Hasler, Jennifer O.
Affiliation:School of Electrical and Computer Engineering, Georgia Institute of Technology, Atlanta, GA, USA
Sound synthesis is often conceptually divided into modules that create or process audio, such as oscillators, filters, and amplifiers, and modules that generate control signals that vary parameters over time, such as pitch of oscillators, cutoff frequency of filters, and gain of amplifiers. Recent advances in field programmable analog arrays (FPAAs) that employ floating-gate transistors have opened the possibility of new analog music synthesizer designs with greater flexibility and higher levels of integration that merge audio processing and parametric control. This report describes implementations of voltage-controlled oscillators (VCOs) and envelope generators on FPAAs using current-starved inverters. VCOs can be used at audio rates or as low-frequency oscillators. Experiments conducted on an FPAA illustrate the modulation of the cutoff frequency of a voltage-controlled filter and the gain of a voltage-controlled amplifier implemented with a Gilbert multiplier.
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The active control of noise levels and sound quality in cars is a pronounced theme in recent research on automotive audio. There’s plenty of processing power available on system-on-chip devices, and this can be used to centralize all of the functions needed. Fast digital buses can be used for transferring audio and control data between elements of the automotive audio system encountered in real situations, reducing the need for large cable bundles.
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