The evaluation of twelve models of loudness perception is presented. One of the loudness models is based on a novel algorithm, and another is based on a combination of two known measurement techniques. The remaining models are all implementations of common or standardized loudness algorithms. The ability of each model to predict or measure the subjective loudness of speech and music segments is evaluated. The reference loudness is derived from two listening experiments using the speech and music segments as stimuli. Different statistical measures are employed in the evaluation of the models, so that both the absolute performance of the models and the performance relative to the between-listener disagreement are measured.

Evaluation of Different Loudness Models with Music and Speech Material

An open area security system comprises an acoustic sensor array (111-1, ..., 111-m) capable of forming elevational and azimuthal beams, or comprises two such arrays separated by a predetermined distance. A camera (101-1, ..., 101-n) mounted in the vicinity of the arrays may be automatically directed toward a detected, sound-producing event. Event data may be prestored in memory (105) and the system may learn of the event's character as an emergency or non-emergency status. Triangulation and other computational techniques may be utilized to determine from the beams the location (x/y coordinates) of the event, thus allowing the camera to be focused and zoomed to capture high resolution images of the event.

Open area security system

A disclosed system and method dynamically controls the perceived volume of a stereo audio program including left and right channel signals. The system comprises: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and an excessive spatial processing protection processor configured and arranged for controlling the level of a difference signal, created as a function of the right channel signal subtracted from the left channel signal (L-R), relative to the level of a sum signal, created as a function of the right channel signal plus the left channel signal;; wherein the excessive spatial processing protection processor processes the audio signals so as to control the difference (L-R) signal relative to the sum (L+R) signal. A system and method are also provided for dynamically controlling the perceived volume of a stereo audio program including left and right channel signals, comprising: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and a program change detector configured and arranged to provide a program change signal indicating that the volume of the left and right channel signals has dropped below a threshold level for at least a threshold time period so as to anticipate a possible change in the sound level of the left and right channel signals; wherein the dynamic volume control is responsive to the program change signal.

Dynamic volume control and multi-spatial processing protection

A single control determines both volume and the degree of compression in the reproduction of motion picture soundtracks. For some settings of the control, compression or limiting reduces the highest levels on a movie soundtrack, leaving the dialogue level substantially unchanged, thus removing the reason for complaints from the audience without the danger of making the dialogue too quiet for intelligibility.

Volume and compression control in movie theaters

Soundfield inside an enclosed space depends in a complex way upon interactions between emitted sound waves and different reflecting, diffracting, and scattering surfaces. 3-D microphone arrays provide tools for investigating and recording these interactions. This paper discusses several existing array techniques introducing a variety of application targets for the HUT microphone probe. Applications include directional measurement, analysis, and visualization of room responses, estimation of room parameters and analysis of source and surface positions. In a dynamic case the probe can be utilized in source tracking and beam steering, as well as in tracking its own position. Furthermore, the probe can be used to simulate some microphone arrays commonly used in surround sound recording. In each application case both general theory and its relation to the HUT probe is discussed. INTRODUCTION In all practical situation soundfield inside an enclosed space consists of acoustical waves propagating in several different directions. Sound waves emitted by a source are reflected, diffracted, and scattered by different obstacles including the walls of the enclosure. This results in a complex field, the properties of which cannot be comprehensively captured in one-dimensional signals or parameters. Different practitioners have different viewpoints into spatial sound. Researchers and acoustical engineers often want to measure a response to a given stimulus in a room, in order to gain information about the reasons why the room sounds like it does. The motivation for this may be an attempt to change the acoustics, or a pure scientific interest in the underlying phenomena, including perception of spatial sound. A recording engineer, on the other hand, may want to capture a performance in a room so that an illusion of the room can be later reproduced somewhere else. Another related problem is selective recording of certain sound sources. Third distinct area of interest is acoustical orientation, i.e., localization of sound sources, reflecting obstacles and surfaces, or the receiver itself based on received sound signals. Of course, there 5501 MERIMAA APPLICATIONS OF A 3-D MICROPHONE ARRAY Fig. 1: The HUT 3-D microphone probe. are also intersections between these viewpoints. This paper describes applications of a 3-D microphone array in all the previously mentioned tasks. The HUT microphone probe consists of 12 miniature electret microphone capsules arranged as two concentric pairs in each of x-, y-, and zcoordinate axes. The inner pairs are set with a spacing of 10 mm and the outer pairs with a spacing of 100 mm between the capsules. The probe was originally designed for measurement purposes but has proven useful in other applications as well. A picture of the probe is shown in Fig. 1. Related hardware and software are described in [1] and [2]. The paper is divided into three parts. The first part discusses directional measurement and analysis of room responses. Second part presents some possibilities of using the probe in sound recording. Finally, the third part introduces source localization techniques applicable both in measurement purposes and in acoustical orientation. ROOM RESPONSE MEASUREMENTS Measurement of room responses and analysis of related attributes is a common task in audio and acoustics. Most often an omnidirectional response to a preferably omnidirectional stimulus is acquired. This is sufficient for calculation of several room-acoustical parameters. However, single omnidirectional responses and standard parameters provide only limited information about the actual acoustics of the room and its perceptual properties. Microphone array techniques utlilized in room response measurements can be roughly divided into two categories. In the first category large arrays spanning a significant distance, area, or volume in the room are utilized. This gives a representation of the evolving soundfield as a function of spatial position. Application of a long line array in this purpose has been described in [3]. In the second category small arrays such as the HUT microphone probe are used to give a listener centered view of the directional soundfield. The following discussion concentrates on methods in the latter category. Directional sound pressure components Ideal omnidirectional microphones are sensitive to sound pressure, which as a scalar quantity does not include any directional information. Systems with varying directional sensitivity can be formed by appropriately combining the signals of two or more closely spaced omnidirectional microphones. An attractive feature of using an array of omnidirectional microphones in place of directional microphones is the possibility to vary and steer the directivity patterns later in the postprocessing phase. First-order differential directivity patterns can be easily created using the signals of a closely spaced pair of microphones. This kind of beamforming methods are analogous to the construction of microphones with built-in directionality [4, 5]. Basically, all that is needed is some equalization and delay, and a weighted summation of the resulting signals. An ideal dipole has a directivity pattern of the form

Applications of a 3-D Microphone Array

An apparatus for monitoring the noise level of passing vehicles and for automatically photographing a vehicle which emits noise that exceeds a predetermined threshold level. The apparatus includes a vehicle sensor which acoustically determines when a passing vehicle reaches a chosen target area. In a preferred embodiment of the invention the vehicle sensor includes a pair of microphones and an acoustic delay line which feeds one of the microphones. The microphone outputs are correlated to determine when the vehicle enters the target area.

Noise monitoring apparatus

Method and apparatus for frequency sensitive amplitude limiting

A loudness limiter which reduces disparity in loudness levels, when a program is controlled on a vu basis, has been developed by CBS Laboratories. The device evaluates the loudness level of the program by a "loudness level summation" method described previously by the authors, and automatically reduces the program level when a preset threshold is exceeded.

Automatic Control of Loudness Level

This paper is a survey and a progress report on present and potential instrumentation to measure loudness levels of sounds. The development of the sound‐level meter (SLM) in the 1930's, its technological and psychoacoustical evolution, and its evaluation as a measurement device are discussed. A loudness‐level monitor, designed to measure the relative loudness level of broadcast programs directly on an indicating instrument, and the development of a new set of equal‐loudness contours are described, together with a new law of loudness‐level addition for combined bands of noise which was discovered during the research. An experimental loudness level meter, devised by adapting CBS Laboratories' loudness level monitor and the potential development of an accurate and practical loudness‐level indicating device are discussed. The results of psychoacoustic evaluations of the loudness level meter versus sound‐level meter are given.

The Measurement of Loudness Level

As described herein, a loudness level indicator for measuring the loudness levels of broadcast (and other) sounds includes an equalizing network having a transfer characteristic which is the inverse of an equal loudness contour representative of the loudness levels of broadcasting sounds over a selected range of pressure levels. Signals corresponding to the broadcast sounds are equalized by the equalizing network such that sounds of any particular frequency having equal loudnesses are provided as voltage signals having equal amplitudes. These equalized signals are separated by a plurality of filters and the separated signals are then combined linearly by a combining network. A ballistics shaping network shapes the combination signals and supplies the combination signals to an indicating instrument which provides a visual indication of the amplitudes of the combination signals and, accordingly, the loudness levels of the broadcast sounds.

Richard G. Allen

Papers

Noise monitoring apparatus

Method and apparatus for frequency sensitive amplitude limiting

Automatic Control of Loudness Level

The Measurement of Loudness Level

Loudness level indicator