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In recent years there has been a good deal of research undertaken on the in-room responses of cinemas, most notably on how they are measured and calibrated. Much of this research has been centered on measurement procedures such as the number of microphone positions to be spatially averaged, the measurement equipment itself and the manipulation of the resultant room curve. This paper proposes that cinema measurement and calibration should instead place increased focus on the anechoic data of the loudspeaker being utilized. In this research, a well-defined loudspeaker was utilized and the playback system was held constant in each cinema so that the interaction between only the loudspeaker and the room could be observed.
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In most professional, commercial and domestic cinemas the loudspeakers are mounted behind the screen so that the visual images on screen coincide with the position of the associated sound images. However, this requires the screen material to be both light reflective and acoustically transparent, which is difficult to achieve. The resultant, imperfect sound transmission gives rise to the sound from the loudspeakers being reflected back from the screen and then forwards again as it reflects from the loudspeaker. The interference between the initial sound and the subsequent reflexions gives rise to comb-filtering which depends upon the reflection coefficients of the screen and loudspeaker and the spacing between the screen and the loudspeaker (see Figures 1 & 2). A change in separation will cause the respective phase difference between the direct and reflected waves to vary, thus resulting in a different resultant wave at the point of the receiver (see Figures 2 & 3). Little research has been carried out to date on the audibility of this comb filtering and in particular, given a certain screen / loudspeaker setup, the effect of varying the spacing has not been investigated thoroughly. By modeling the system in Matlab, and using existing experimental data as a comparative reference, the behavior of the perforated screen when subjected to an incident sound wave can be predicted. Once this is achieved, the effective filter for different separations between screen and loudspeaker can be used to create a database of simulated output signals. These signals represent a portion of cinema audio being played through the screen / loudspeaker system in different geometric setups. These signals can then be used to determine the audibility of the comb filtering effect at different separations via a comparative subjective testing technique.
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Frequency responses and RTA measurement results of cinema screen systems are presented for a small selection of cinemas sizes and shapes in Australia. Measurements have been made using time gated techniques (MLSSA) and the traditional pink noise Real Time Analyser (RTA) process with a microphone array. Results from these methods will be compared with the manufacturer’s quasi anechoic measurements, and probable reasons for the differences will be discussed.
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Perforated cinema screens are currently in widespread use in cinemas and dubbing stages, due to their high light-reflectivity. However, in the acoustical domain, such screens form a low pass filter which attenuates the high frequency response of the cinema loudspeakers. Recent studies have shown that the X-curve, which has long been adopted by the SMPTE as standard responses curve, primarily reflects this low pass filter action. It is therefore of significance to explore the ramifications of equalising the effects of this low pass filter, in order to provide a flat frequency response for listeners. This paper reports on measurements of the spectral content of several movies, and assesses the impacts on the headroom in the playback chain and the long-term power requirements of the loudspeaker drivers that equalisation of this type would impose with the spectral content of these movies.
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This paper explores strategies for achieving accurate wide-area low-frequency sound reproduction in cinemas. Current standards for B-Chain calibration call for single channel low-frequency equalization aided by either single-point or spatially-averaged response measurements, an approach only applicable to a reasonably spatially invariant low-frequency response. A holistic approach to low-frequency coverage optimization is presented exploiting subwoofer arrays, their positioning and multi-point signal processing. Acoustic-field examples are presented using finite-difference time-domain (FDTD) modeling software that expose a potential for superior wide-area signal reconstruction over that achieved using the current standards and recommendations.
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Little has been published about the repercussions of different source locations and measuring positions for the Low-Frequency Effects (LFE) loudspeakers in cinemas and dubbing theatres. The aim of this study is to determine the effects of the number and position of the loudspeakers on the uniformity of the response over the listening area, and to assess the effect of the measuring of those responses by the choices made regarding the positioning of the microphones within typically used arrays. Some organisations issuing recommendations for the installation of LFE loudspeakers have long suggested that they should be installed asymmetrically, below the screen. The reason given for this is to avoid the symmetrical driving of the low-frequency modal responses of the rooms, the sources typically being centred about 20% of the distance from one side-wall and 33% of the distance from the opposite side wall. However, other organisations, and many designers, have recommended the mounting of the LFE loudspeakers in tighter-packed clusters. Furthermore, it has long been suggested that a single microphone position is inadequate for measuring the response of an LFE channel because the long wavelengths involved make spacial variation in the measurements inevitable. Typically, 4, 5, 8 and 10-microphone arrays have been used, but as rooms differ so much in size, shape and acoustic properties, no universal instructions exist about precisely how to place the microphones in such arrays. This paper examines how the choice of low-frequency source positions, and the microphone positions in a 5-microphone array, can affect the measured average responses over the designated listening areas. It also discusses how the positioning of the loudspeakers can affects the evenness of coverage. The results of the spacially averaged responses are compared with the responses at individual microphone positions, in order to assess how representative the averages are of the responses at specific locations.
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The loudness, dynamic range and energy distribution in low-frequency bands of popular music are analyzed. One objective was to operationalize popular music and construct a robust, balanced sample that covers a specific but relevant music market regarding annual revenues. The sample consists of the “German Top 40” year-end charts from 1965 to 2013. Furthermore, different methods of measurement, such as LKFS or dBFS RMS, are used and compared. It could be shown that there was a significant increase of loudness, a decrease of the dynamic range and an increasing importance of the low-frequency bands over time. While our results correspond to most previous research, there is a major difference regarding the recent data. It is frequently mentioned in studies that the process of decreasing dynamic range peaked in 2004, and after that the opposite trend occurred, namely, an increase in dynamic range. In the German music market, however, this seems to be true only for the time span from 2004 to 2010. From 2011 to 2013 a significant decrease of the dynamic range and an increase in loudness were found.
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Audio quality is very important to broadcasters’ audiences, and unwanted loudness variations are a source of complaints. Dynamic range control applied by the broadcaster can go some way to avoiding problems, but this has numerous problems, one being that not everyone wants the same thing. The web audio API provided by HTML5 offers the possibility of performing dynamic range control under the control of the listener, tailoring it optimally for their individual situation. We have produced a system that demonstrates clearly that this is simple to achieve in a modern web browser, automating control of the compressor using environmental noise level measured using the microphone present in most mobile device audio players.
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In this paper we investigate the perception of sound source distance in relation to static and dynamic binaural systems. Reference data for distance perception in the median plane is first presented, which shows through subjective evaluation that under the test conditions there is no perceived difference in distance perception with sound sources presented at different azimuthal angles to the head. This data forms the hypothesis that head movements do not play an important role in auditory perception of source distance in real rooms. This is verified in relation to binaural systems through comparison of distance perception of binaural recordings versus head-tracked binaural rendering. The results also demonstrate that non-individualised head-related transfer functions can be effective for distance perception when compared to real sources.
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This paper describes an experiment designed to study the differences between 5.1 audio played through loudspeakers and headphones in binaural, and between compressed, and uncompressed audio files. Differences in terms of spatial impression and of overall quality of the sound have been studied. This experiment was made in the context of 'NouvOson', a Radio France website launched in March 2013 (http://nouvoson.radiofrance.fr), where audio contents are available online both in native 5.1 and processed in binaural using SpherAudio software by Digital Media Solutions. It also concerned the BILI Project, dealing with BInaural LIstening, involving Radio France, France Televisions and DMS. Binaural processing theoretically allows the reproduction of 3D sound when listening through headphones; however, this technology still faces issues. These are not only due to the actual limits of research and development, but also to the way we listen to and localize sounds. This experiment has shown that spatial charasteristics, as well as timbre of the sound are modified. Besides, no real difference in the listener's perception has been found between binaural uncompressed files and AAC 192 kbps as well as MP3 192 kbps files.
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