Multichannel Reproduction Research Articles

Application of sound field reproduction system to acoustic welfare engineering - Real time sound field transmission system for parent and child room in a concert hall -

This study attempted to implement a real-time sound field reproduction system for a parent-child (PC) room in a concert hall as an example of acoustic welfare engineering. The PC room is typically located at the rear of a hall where visitors with their children can enjoy performances without hesitation, but the sound quality is rarely discussed. This study attempts to improve this situation using a multi-channel reproduction system. Specifically, we implemented a system in which the directional impulse response of the hall measured by a microphone array is convolved in real-time with the sound played and captured separately on stage and reproduced by a loudspeaker array installed in a PC room. The 24-channel system, in which the polar and azimuth angles are both 45 degrees apart, our standard procedure, was found to have a high degree of physical reproducibility. Still, it needed to be more practical considering poor room conditions such as large windows. Therefore, we introduced panning using loudspeakers above and below the window frame instead of using the front loudspeakers. The impulse responses of the original and reproduced sound fields were compared, and it was found that the reproducibility was maintained even with the introduction of the panning.

A listener preference model for spatial sound reproduction, incorporating affective response.

This work presents results and models for listener preference to music delivered via different spatial reproduction formats, here via mono, stereo and multichannel (5.1-ch) reproduction. Although this problem has been researched in the past, the current work introduces an elaborate multistage experimental procedure which considers the contribution of listener-specific emotional responses (valence and arousal) to his/hers Overall Listening Experience (OLE). The test procedure registers the individual listener preference and familiarization with the content of each test audio sample. A spatial envelopment metric, extracted directly from each audio signal sample is utilized as attribute for the perceived differences between the 3 different systems. This attribute, along with listener content preference (for each music sample) and his/hers affective response attributes are combined into linear regression model(s) which can predict the dominant trends for rating OLE. A novel linear tree approach is also proposed which highlights additional associations between the attributes within this multidimensional space. Comparative performance analysis shows that the proposed linear tree approach also achieves improved predictions for OLE ratings.

The Soundlapse Project: Exploring Spatiotemporal Features of Wetland Soundscapes

Abstract This article discusses an interdisciplinary project aimed at highlighting the acoustical heritage of wetlands by means of field recordings and a novel time-lapse montage method. The authors discuss a site-specific sound installation that was designed using original wetlands field recordings, live processing and spatial audio multichannel reproduction. The discussion focuses on spatial and temporal features of different types of recorded wetlands soundscapes. Future developments of this project will consider the implementation of a standalone spatiotemporal application, to be used in the context of virtual reality applications, game audio and interactive dance performance.

Late-Reverberation Synthesis Using Interleaved Velvet-Noise Sequences

This paper proposes a novel algorithm for simulating the late part of room reverberation. A well-known fact is that a room impulse response sounds similar to exponentially decaying filtered noise some time after the beginning. The algorithm proposed here employs several velvet-noise sequences in parallel and combines them so that their non-zero samples never occur at the same time. Each velvet-noise sequence is driven by the same input signal but is filtered with its own feedback filter which has the same delay-line length as the velvet-noise sequence. The resulting response is sparse and consists of filtered noise that decays approximately exponentially with a given frequency-dependent reverberation time profile. We show via a formal listening test that four interleaved branches are sufficient to produce a smooth high-quality response. The outputs of the branches connected in different combinations produce decorrelated output signals for multichannel reproduction. The proposed method is compared with a state-of-the-art delay-based reverberation method and its advantages are pointed out. The computational load of the method is 60% smaller than that of a comparable existing method, the feedback delay network. The proposed method is well suited to the synthesis of diffuse late reverberation in audio and music production.

Хоровое письмо Карлхайнца Штокхаузена в гепталогии «Licht»

The article is devoted to the phenomenon of Karlheinz Stockhausen’s choral writing in his large- scale opera project Licht. The musical composition’s religious-philosophical basis, which initiates the transformation of the opera theater into a quasi-temple space, leads to the reinforcement of the dramaturgical role of choral music, which manifests itself in the intensification of the presence of the chorus in the opera project (massive choral scenes appear in all the heptalogy’s operas) and in the expansion of its function. The diversity of the types of exposition of the musical material, nonetheless, makes it possible to disclose as a general tendency the aspiration towards choral stereophony, both in the situation of a multichannel reproduction of previously recorded choral material (Unsichtbare Chöre) and in live performance (practically all the choral scenes in the heptalogy). Special attention is given to analysis of the verbal basis of the choral scenes. Along with incorporating his own texts, Stckhausen turns to the sacred literature of Christianity, Judaism, Hinduism and Islam. Combination of poly-lingual texts in different styles in the conclusive scene of the Hoch-Zeiten symbolizes the spiritual relatedness of all of humanity. An important technical resource is the use of elements of phonemic composition, exerting direct influence on the choral timbres. The composition’s timbral side attracts attention to it by its connection together of elements of vocal technique already used by Stockhausen in his preceding compositions (4 modifications of vocal tremolo, glissando, phonic intonating, Sprechgesang, singing while inhaling, tongue clicks, etc.) with special techniques typical particularly for the score of Licht. The latter include “color noise” (“fabriges Rauschen”), yodeling and a “kissing sound” (“Kußgeräusche”). The organic quality of such a complex is guaranteed by the unity of the conception of Stockhausen’s heptalogy, which the exerted influence not only on the opera cycle’s compositional technique, but also on its scenography and dramaturgical solutions. Keywords: Karlheinz Stockhausen, heptalogy Licht, contemporary choral music, texture, stereophony, choral timbre, music and the word, phonemic composition.

Reflections on sonic digital unreality

ABSTRACTThis paper explores the possibility of constructing an unreal environment through sound, particularly through the use of contemporary digital tools. The idea of the unreal is approached as a not being-in-the-world experience. That is, an experience which expands the natural embodied physicality. And yet, sound propagation through a physical space, which led to the study of reverberation and spatialization as the main subjects of interest. Consequently, reverb became the main tool to explore the subject in practice. Due to the project’s multidisciplinarity, the subject had different approaches: from the possibility of reverb as a tool for fiction-making, to listener's comprehensibility and readability of the events; exploring spatialization techniques and multichannel reproduction systems to finally conclude with the idea that, in the end, sound relies on the listener’s actual physical experience of the space.

Analysis on decorrelation effect of audio signal in multichannel reproduction

Decorrelation of audio signal is a process that generates two or more incoherent signals from a single input signal, which has many applications in artificial auditory effects, such as broadening the apparent source width (ASW), enhancing the subjective envelopment, producing subjective diffusion in multichannel reproduction, etc. Frequently used decorrelation method is convoluting audio signal with random (or pseudo-random) noise. In this work, we focus on the relationship of reproduction zone of multichannel reproduction with decorrelation degree. Simulation calculation result show that, by changing the decorrelation degree which is measured by the interaural cross-correlation coefficients (IACC) of reproduced signals, the timbre will change when move off the listening center. Listening zone with small timbre deviation (for example, binaural loudness spectra less than 1dB/ERB) relative to center position increase with the value of IACC decrease. But as well known, for good subjective diffusion, the least IACC value does not mean the best effect. There exit a best IACC value range from 0.2~0.4 for different audio signal. Adapting a best IACC value, a reproduction zone across different frequencies can be achieved.

Multichannel Acoustic Echo Cancellation in the Wave Domain With Increased Robustness to Nonuniqueness

Acoustic echo cancellation (AEC) is significantly more challenging in the case of multichannel reproduction compared to the single-channel case. This is mainly due to the so-called nonuniqueness problem, which becomes especially severe for massive multichannel reproduction systems like, e.g., wave field synthesis systems or higher-order ambisonics, where the different loudspeaker signals are often strongly correlated. In this paper, a multichannel wave-domain AEC structure is proposed that maintains robustness without altering the loudspeaker signals, as is commonly required to preserve the reproduction quality. The proposed method exploits the dominance of certain couplings in the wave-domain representation of the loudspeaker-enclosure-microphone system. In this representation, preferring a strong weight of the dominant couplings in the identified system improves the system identification performance. A modification of the well-known generalized frequency-domain adaptive filtering algorithm is then presented to implement the proposed approach, including a self-contained description of all necessary components. An experimental evaluation compares the proposed method to competing approaches and shows that the proposed method compares well in terms of convergence behavior while avoiding the performance degradation, which is intrinsic to other approaches.

The generalized frequency-domain adaptive filtering algorithm as an approximation of the block recursive least-squares algorithm

Acoustic echo cancellation (AEC) is a well-known application of adaptive filters in communication acoustics. To implement AEC for multichannel reproduction systems, powerful adaptation algorithms like the generalized frequency-domain adaptive filtering (GFDAF) algorithm are required for satisfactory convergence behavior. In this paper, the GFDAF algorithm is rigorously derived as an approximation of the block recursive least-squares (RLS) algorithm. Thereby, the original formulation of the GFDAF algorithm is generalized while avoiding an error that has been in the original derivation. The presented algorithm formulation is applied to pruned transform-domain loudspeaker-enclosure-microphone models in a mathematically consistent manner. Such pruned models have recently been proposed to cope with the tremendous computational demands of massive multichannel AEC. Beyond its generalization, a regularization of the GFDAF is shown to have a close relation to the well-known block least-mean-squares algorithm.

The preferred level balance between direct, early, and late sound in concert halls.

The preferred level balance between the direct, early, and late components of the sound field in a concert hall was studied by letting listeners manipulate sound field auralizations in an adjustment procedure featuring acoustics of two halls of contrasting designs—Berlin Konzerthaus (shoebox) and Berlin Philharmonie (vineyard). The room impulse responses measured at two positions in each hall were encoded for multichannel reproduction, split into direct, early, and late components, convolved with anechoic recordings of a symphony orchestra, and reproduced in a multichannel listening room. The listeners’ task consisted of optimizing the level balance between the direct, early, and late components by adjusting the level of two of the components relative to one that was kept fixed. The results reveal a strong tendency to emphasize the early component for both positions in Berlin Philharmonie, where the early reflections contain less energy due to the fan-shaped frontal auditorium and comparatively small effective side wall areas. In contrast, the median results for one position in Berlin Konzerthaus were within 1.5 dB of the actual acoustics of the hall, with statistically insignificant deviations. The relationship between the direct sound and early reflections is seen as a potential subject of further study.

Auralization of virtual rooms in real rooms using multichannel loudspeaker reproduction

When playing auralizations including virtual room reverberation through loudspeaker-based reproduction systems, there is usually an interaction between the auralized virtual rooms with the real room acoustics of the listening environment. In case of a listening room which is not perfectly dry, it is investigated which criteria the listening room should fulfill to avoid considerable interference with the auralizations. In a further step, a computer room acoustics simulation is extended to account for the listening space by modifying the resulting room impulse responses, so that the final room-in-room situation matches best to the targeted virtual room acoustics. The presented technique is then applied in a multimodal immersive virtual display (CAVE-like environment) where room acoustics are not matter of choice due to restrictions of projection screen materials and placement.

Large-scale multiple input/multiple output system identification in room acoustics

In audio reproduction scenarios, room acoustics may be described as a MIMO system response from multiple loudspeaker to multiple microphones in the listening space. This system response may, e.g., be used for an equalization of the listening room and must be identified from observing the available loudspeaker and microphone signals in real-world systems. For few transducers this task is mostly solved, but massive multichannel reproduction with dozens to hundreds of loudspeakers left many research questions open. This contribution points out the fundamental challenges, previous solutions and recent advances. As a key issue, the so-called nonuniqueness problem for MIMO system identification by adaptive filtering will be discussed along with decorrelation schemes for the loudspeaker signals to alleviate this problem. Successful adaptation algorithms suitable for these scenarios imply considerable computational demands and require additional measures to ensure robustness. Recently emerging system models in spatial transform domains allow for approximative models and seem to be promising for robust real-time implementations.

Auralization of municipal public address announcements by applying geometrical sound simulation and multi-channel reproduction techniques

Municipal public address system (M.P.A. system) for disaster prevention is an important system for supplying information outdoors. The speech intelligibility of the M.P.A. system, however, tends to be deteriorated by long-pass-echoes due to reflections from the surrounding buildings and the sounds from the loudspeakers covering other sub-areas. When designing such a M.P.A. system, effective tools for prediction of outdoor sound propagation are needed. For this aim, the authors have been investigating the application of the sound simulation technique based on geometrical acoustics. In addition, auralization is also desirable to subjectively assess the speech intelligibility of the M.P.A. system in acoustical design. Therefore, the authors are developing a simulation technique in which the sounds calculated by the geometrical simulation are reproduced through a 6-channel reproduction system, by which 3D information can be aurally realized. In this paper, the outline of this simulation method and some exampl...

Method for Reproducing Natural or Modified Spatial Impression in Multichannel Listening

The invention concerns a method for reproducing spatial impression of existing spaces in multichannel or binaural listening. It consists of following steps/phases: a) Recording of sound or impulse response of a room using multiple microphones, b) Time- and frequency-dependent processing of impulse responses or recorded sound, c) Processing of sound to multichannel loudspeaker setup in order to reproduce spatial properties of sound as they were in recording room, and (alternative to c), d) Processing of impulse response to multichannel loudspeaker setup, and convolution between rendered responses and an arbitrary monophonic sound signal to introduce the spatial properties of the measurement room to the multichannel reproduction of the arbitrary sound signal, and is applied in sound studio technology, audio broadcasting, and in audio reproduction.

Evaluating urban soundscape quality with sound recording and reproduction.

Sound recording is an obvious method when the sound quality of a soundscape needs to be assessed. The benefit of a recording, beyond archival purposes, is that the signal can be analyzed to obtain objective metrics of quality such as tonality, spectral composition, roughness, fluctuation strength, etc., and it may also be played back to listeners under controlled settings for the purpose of qualitative evaluation. The goal of the research reported here was to evaluate the advantages and disadvantages of different recording and reproduction methods and determine which method offered the greatest correlation between subjective evaluations made in a laboratory and those made \emph{in situ}. In addition, binaural and multichannel reproduction techniques were explored in tandem with photographs and videos to create different levels of immersion and determine the impact of visual cues on sound quality evaluation. Two parks in Manhattan were chosen as test environments for this study. All participants were given identical surveys which evaluated sound quality using descriptive content analysis techniques and semantic differential ratings.

Localization of virtual sources in multichannel audio reproduction

The localization of virtual sources generated with different two-dimensional (2-D) multichannel reproduction systems has been studied by means of auditory model simulations and listening tests. The reproduction was implemented with typical five- and eight-channel loudspeaker setups. The microphone systems used were first- and second-order Ambisonics as well as a spaced microphone technique. Pair-wise panning was also studied. The results show that the auditory model can be used in the prediction of perceived direction in multichannel sound reproduction near the median plane. Some systematic deviations between the model predictions and the listening test results were found farther from the median plane. The frequency-dependent capability to produce narrow-band virtual sources to targeted directions is reported for the studied systems.

Introduction to the Special Issue on Multichannel Signal Processing for Audio and Acoustics Applications

HE IEEE Signal Processing Society has its roots in an area where acoustics, speech, and signal processing converge, as was reflected in the former name of the society when it was founded in 1974. The interface between acoustics, speech, and signal processing is still an area of great interest to the society, with many fundamental problems still unsolved. Research is driven by applications where acoustic signals have to be captured, transmitted, and/or reproduced in an acoustic environment that includes echoes, noise, and reverberation Considering human/machine interfaces as a major area of applications, it is obvious that signal processing becomes more challenging as the distance between humans and the machines increases, as the signal bandwidth increases, and as the acoustic environment becomes more complex and hostile. Increasingly sophisticated algorithms have been developed since the mid-1970s and along with the availability of greatly increased and affordable computational power, multichannel signal processing algorithms naturally evolved for exploiting the spatial dimension of acoustic signals. The importance and popularity of this field was well reflected by the large number of submissions to this special issue. The volume of high-quality papers could not be fitted into the page budget allotted to us. Thus, we regrettably had to decide to publish some of them in a second section of this special issue as part of a regular issue of the TRANSACTIONS in early 2005. For sound reproduction, where we want to provide a pair of desired signals at the listeners’ ear drums, seamless human/machine interfaces based on multichannel techniques have been implemented since the invention of stereo systems. However, providing the true spatial sound experience in large listening spaces became possible only with new multichannel signal processing techniques, such as wavefield synthesis. Still, major challenges remain, especially phase-true equalization of listening room acoustics and the cancellation of local noise sources and interferers. On the other hand, acquisition of audio and speech signals has been a research topic since the invention of the microphone and still today presents major challenges for the signal processing community. Structurally the simplest problem, the acoustic feedback from loudspeakers into microphones is addressed by acoustic echo cancellation: From the single-channel case which has been investigated since the 1970s, research has moved on to stereo and multichannel reproduction, recently culminating in a new wave-domain adaptive filtering concept which has been presented for the first time at ICASSP 2004. For removing unwanted interference and noise from desired signals, multichannel techniques utilize spatial diversity to discriminate between desired and undesired components, either by exploiting different spatial coherence properties or by beamforming, which directs a beam of increased sensitivity towards the desired source. For traditional beamforming, source localization is necessary if the location of the source is not known a priori. Reflecting current research emphasis in these areas, the paper by Cohen addresses filtering of the

Sound field simulation method by combining finite difference time domain calculation and multi-channel reproduction technique

In order to auralize the room impulse responses obtained by the numerical analysis based on the wave theory, a multi-channel sound field simulation system using the Finite Difference Time Domain (FDTD) method has been developed. In this system, uni-directional impulse responses for four orthogonal directions in a 2-dimensional space are firstly calculated by the FDTD method and they are reproduced directly from four loudspeakers set in an anechoic room. At the center point of the reproduced sound field, we can hear the sound with 2-dimensional information assumed in the FDTD calculation. In this paper, the principle and the basic performance of this system are introduced and an example of subjective hearing test performed using this system is presented.

Time-axis compression/expansion method and apparatus for multichannel signals

A time-axis compression/expansion method and apparatus for multitrack signals is provided, which is capable of performing time-axis compression/expansion on a multitrack signal in such an appropriate manner as to prevent a degradation in the sound quality of a sound generated through a multichannel reproduction or a sound generated through reproduction of a musical tone signal obtained by mix-down. Positions of attacks of the rhythm track sound source signal of a plurality of track sound source signals are detected. Portions of the rhythm track sound source signal between the detected positions of attacks are subjected to a first time-axis compression/expansion process, and the other track sound source signals are subjected to a second time-axis compression/expansion process, based on the detected positions of attacks.

Acoustics and psychoacoustics of desk-top sound systems

Sound on the desk top has some similarities and some differences from reproduction over other standardized sound systems such as those employed in the production of film and television programs. This work examines areas where the application of known acoustic and psychoacoustic principles may be used to produce a system that scales the experience of large-scale sound systems optimally to the desk-top environment. Among the methods employed are multichannel reproduction, direct-field dominant reproduction over a wide frequency range, optimal equalization for the conditions encountered by such systems, and wide frequency and dynamic ranges, among others to be discussed. Research topics underway include methods for head tracking and optimal control over the sound field for one listener, with the eventual goal of eliminating surround loudspeakers and replacing them with virtual loudspeakers. The current finding is that contemporary systems are excessively sweet-spot dependent and change the timbre too much to be acceptable professionally in this application, so the direction of research is to reduce these limitations to make a practical system with just three loudspeakers (left, right, and subwoofer).