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J. B. Moore - 2019 - Dynamic Diffuse Signal Processing in Sound Reinforcement and Reproduction
Abstract and Figures
High inter-channel coherence between signals emitted from multiple loudspeakers can cause undesirable acoustic and psychoacoustic effects. One example is commonly known as “comb-filtering”, which occurs when time delayed coherent signals sum. In sound reinforcement, comb-filtering is perceived by audience members as variations in magnitude response which depend upon their position in the audience area.
This situation is the antithesis of a key objective of the sound reinforcement system designer – that all audience members should, within reason, receive a signal that is faithful to the source material.
Nevertheless, the problem is present in all sound reinforcement scenarios that employ multiple loudspeakers emitting coherent signals to meet audience coverage and sound pressure level (SPL) needs.
Typically, low frequency sound reproduction is handled by a dedicated array of distributed
loudspeakers, each emitting an identical signal (with the possible exceptions of electronic time delay and polarity). Because of the size of audience areas in large-scale sound reinforcement scenarios, it is possible for audience position to loudspeaker path-length differences to be on the order of meters [21], leading to significant differences in the time of arrival of contributing signals at a given audience location. The result of this is positionally dependant comb-filtering that extends to the low frequency range.
This work seeks to develop a signal processing algorithm that addresses this issue by lowering the inter-channel coherence of distributed loudspeaker arrays so that the summation of their outputs is no longer dependent on their relative phases. This means that regardless of path-length differences, the summation of multiple coherent sound sources results in comparable magnitude responses across audience areas.
Initially, the work investigates the suitability of a method of signal decorrelation termed diffuse signal processing (DiSP), which utilises FIR decorrelation filters termed temporally diffuse impulses (TDIs). Several TDI optimisation methods are described, and the performance of the algorithm is examined using an image-source (IS) acoustic model. Whilst DiSP is found to be successful in reducing magnitude response variation across audience areas in non-enclosed acoustic spaces, in enclosed acoustic spaces the performance of the algorithm is reduced.
This reduction in performance is the result of early reflection/direct source summation. The non time-variant nature of DiSP means that whilst inter-channel coherence is successfully reduced, the output of a single discrete source still maintains high correlation with itself over short time-frames. Therefore, when early reflections sum with their direct source’s output, comb-filtering is produced.
A variation of DiSP termed dynamic DiSP is described, which can reduce the correlation of a discrete source’s output from one time instant to the next. This aspect of the processing mitigates comb-filtering that is a product of direct source/early reflection summation. It is also a potentially useful tool in reducing modal behaviour in acoustically small spaces. These claims are investigated in detail in both simulated and real-world scenarios.
A number of user-variables are described which enable easy control of the algorithm, allowing users to balance the level of decorrelation performance required versus the audibility of the processing. Suitable values for these variables are proposed and their perceptual transparency is investigated using a MUSHRA style subjective test. In this test, the algorithm scored 87.3/100 versus an unprocessed signal score of 91.1/100, showing that dynamic DiSP may be applied in a perceptually transparent manner.
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The general aim of live sound reinforcement is to deliver an appropriate and consistent listening experience across an audience. Achieving this in the subwoofer range (typically between 20-100 Hz) has been the focus of previous work, where techniques have been developed to allow for consistent sound energy distribution over a wide area. While this provides system designers with a powerful set of tools, it brings with it many potential metrics to quantify performance. This research identifies key indicators of subwoofer system performance and proposes a single weighted metric to quantify overall performance. Both centrally-distributed and left/right configurations are analyzed using the new metric to highlight functionality.
A central goal in small room sound reproduction is achieving consistent sound energy distribution across a wide listening area. This is especially difficult at low-frequencies where room-modes result in highly position-dependent listening experiences. While numerous techniques for multiple-degree-of-freedom systems exist and have proven to be highly effective, this work focuses on achieving position-independent low-frequency listening experiences with a single subwoofer. The negative effects due to room-modes and comb-filtering are mitigated by applying a time-varying decorrelation method known as dynamic diffuse signal processing. Results indicate that spatial variance in magnitude response can be significantly reduced, although there is a sharp trade-off between the algorithm's effectiveness and the resulting perceptual coloration of the audio signal.
High inter-channel coherence between signals emitted from multiple loudspeakers can cause undesirable acoustic and psychoacoustic effects. Examples include position-dependent low-frequency magnitude response variation where comb-filtering leads to the attenuation of certain frequencies dependent on path length differences between multiple coherent sources, lack of apparent source width in multi-channel reproduction, and lack of externalization in headphone reproduction. This work examines a time-variant, real-time decorrelation algorithm for the reduction of coherence between sources as well as between direct sound and early reflections, with a focus on minimization of low-frequency magnitude response variation. The algorithm is applicable to a wide range of sound reinforcement and reproduction applications, including those requiring full-band decorrelation. Key variables that control the balance between decor-relation and processing artifacts such as transient smearing are described and evaluated using a MUSHRA test. Variable values that render the processing transparent while still providing decorrelation are discussed. Additionally, the benefit of transient preservation is investigated and is shown to increase transparency.
In this paper we propose a new low complexity method of transient detection using an image edge detection approach. In this method, the time-frequency spectrum of an audio signal is considered as an image. Using appropriate mapping function for converting energy bins into pixels, audio transients correspond to rectilinear edges in the image. Then, the transient detection problem is equivalent to an edge detection problem. Inspired by standard image methods of edge detection, we derive a detection function specific to rectilinear edges that can be implemented with a very low complexity. Our method is evaluated in two practical audio coding applications, in replacement of the SBR transient detector in HEAAC+ V2 and in the stereo parametric tool of MPEG USAC.
The use of active crossovers is increasing. They are used by almost every sound reinforcement system, and by almost every recording studio monitoring set-up. There is also a big usage of active crossovers in car audio, with the emphasis on routing the bass to enormous low-frequency loudspeakers. Active crossovers are used to a small but rapidly growing extent in domestic hifi, and I argue that their widespread introduction may be the next big step in this field. The Design of Active Crossovers has now been updated and extended for the Second Edition, taking in developments in loudspeaker technology and crossover design. Many more pre-designed filters are included so that crossover development can be faster and more certain, and the result will have a high performance. The Second Edition continues the tradition of the first in avoiding complicated algebra and complex numbers, with the mathematics reduced to the bare minimum; there is nothing more complicated to grapple with than a square root. New features of the Second Edition include: More on loudspeaker configurations and their crossover requirements: MTM Mid-Tweeter-Mid configurations (The d’Appolito arrangement) Line arrays (J arrays) for sound reinforcement Frequency tapering Band zoning Power tapering Constant-Beamwidth Transducer (CBT) loudspeaker arrays More on specific sound-reinforcement issues like the loss of high frequencies due to the absorption of sound in air and how it varies. Lowpass filters now have their own separate chapter. Much more on third, fourth, fifth, and sixth-order lowpass filters. Many more examples are given with component values ready-calculated Highpass filters now have their own separate chapter, complementary to the chapter on lowpass filters. Much more on third, fourth, fifth, and sixth-order highpass filters. Many more examples are given with component values ready-calculated A new chapter dealing with filters other than the famous Sallen & Key type. New filter types are introduced such as the third-order multiple feedback filter. There is new information on controlling the Q and gain of state-variable filters. More on the performance of crossover filters, covering noise, distortion, and the internal overload problems of filters. The chapter on bandpass and notch filters is much extended, with in-depth coverage of the Bainter filter, which can produce beautifully deep notches without precision components or adjustment. Much more information on the best ways to combine standard components to get very accurate non-standard values. Not only can you get a very accurate nominal value, but also the effective tolerance of the combination can be significantly better than that of the individual components used. There is no need to keep huge numbers of resistor and capacitor values in stock. More on low-noise high-performance balanced line inputs for active crossovers, including versions that give extraordinarily high common-mode rejection. (noise rejection) Two new appendices giving extensive lists of crossover patents, and crossover-based articles in journals. This book is packed full of valuable information, with virtually every page revealing nuggets of specialized knowledge never before published. Essential points of theory bearing on practical performance are lucidly and thoroughly explained, with the mathematics kept to an essential minimum. Douglas’ background in design for manufacture ensures he keeps a very close eye on the cost of things.
A thorough investigation was made of the influence of a single echo as a function of different parameters on the audibility of speech. An apparatus was build for the artificial generation of echoes and measurements were made with a large number of observers under specified conditions.
Since Békésy's excellent experimental measurements of the mechanical constants and movement of the basilar membrane, two major papers have become available which give theoretical discussions of the subject discussed in this paper, one by Peterson and Bogart and one by Zwislocke. In the former the frictional force was neglected and a solution was obtained which throws considerable light on the mechanism, but the calculated curves of amplitude at various positions along the basilar membrane are far from those observed by Békésy. In the latter paper the frictional term is considered the all‐important one, and the vibrating mass is neglected, and the author claims that his calculated amplitude curves fit those observed except for the phases; however, all of the details of this calculation are not given. In the present paper all three terms in the mechanical impedance of the basilar membrane are taken into account, and calculated amplitudes are obtained which agree approximately with those observed in amplitude and phase and time of travel of the wave from the stapes to any point on the membrane.