javax.media.j3d

Class AuralAttributes

java.lang.Object

javax.media.j3d.SceneGraphObject

javax.media.j3d.NodeComponent

javax.media.j3d.AuralAttributes

public class AuralAttributes
extends NodeComponent

The AuralAttributes object is a component object of a Soundscape node that defines environmental audio parameters that affect sound rendering. These attributes include gain scale factor, atmospheric rolloff, and parameters controlling reverberation, distance frequency filtering, and velocity-based Doppler effect.

Attribute Gain

Scale factor applied to all sound's amplitude active within this region. This factor attenuates both direct and reflected/reverbered amplitudes. Valid values are >= 0.0

Attribute Gain Rolloff

Rolloff scale factor is used to model atmospheric changes from normal speed of sound. The base value, 0.344 meters/millisecond is used to approximate the speed of sound through air at room temperature, is multipled by this scale factor whenever the speed of sound is applied during spatialization calculations. Valid values are >= 0.0. Values > 1.0 increase the speed of sound, while values < 1.0 decrease its speed. A value of zero makes sound silent (but it continues to play).

Auralization

Auralization is the environmental modeling of sound iteratively reflecting off the surfaces of the bounded region the listener is in. Auralization components include early, distinct, low-order reflections and later, dense, higher-order reflections referred to as reverberation. These reflections are attenuated relative to the direct, unreflected sound. The difference in gain between direct and reflected sound gives the listener a sense of the surface material and the relative distance of the sound. The delay between the start of the direct sound and start of reverberation (as detected by the listener), as well as the length of time reverberation is audible as it exponentially decays, give the listener a sense of the size of the listening space.

In Java3D's model for auralization there are several parameters that approximate sound reflection and reverberation for a particular listening space:

Reflection Coefficient
Gain attenuation of the initial reflections across all frequencies.
(Early) Reflection Delay
The time it takes for the first low-order reflected sound to reach the listener.
Reverb Coefficient
Gain attenuation of the late reflections (referred to as 'reverb') across all frequencies.
Reverb Delay
The time it takes for reverbered sound to reach the listener.
Decay Time
Describes the reverb decay curve by defining the length of time reverb takes to decay to effective zero.
Decay Filter
High-frequencies of the late reverberation can be attenuated at a different rate.
Density
Modal density (spectral coloration) of reverberation.
Diffusion
Echo dispersement of reverberation.
Reverb Bounds
Approximates the volume of the listening space. If specified, it defines the reverberation delay.
Reverb Order
Optionally limits the amount of times during reverb calculation that a sound is recursively reflected off the bounding region.

Reflection Coefficient

The reflection coefficient is an amplitude scale factor used to approximate the average reflective or absorptive characteristics for early reflections of the composite surfaces in the region the listener is in. This scale factor is applied to the sound's amplitude regardless of the sound's position. The range of valid values is 0.0 to 1.0. A value of 1.0 denotes that reflections are unattenuated - the amplitude of reflected sound waves are not decreased. A value of 0.0 represents full absorption of reflections by the surfaces in the listening space (no reflections occur thus reverberation is disabled).

Reflection Delay

The early reflection delay time (in milliseconds) can be explicitly set. Well-defined values are floats > 0.0. A value of 0.0 results in reverberation being added as soon as possible after the sound begins.

Reverberation Coefficient

The reverb coefficient is an amplitude scale factor used to approximate the average reflective or absorptive characteristics of late reflections. A value of 0.0 represents full absorption of reflections by the surfaces in the listening space (no reflections occur thus reverberation is disabled).

Reverberation Delay

The reverb delay time (in milliseconds) is set either explicitly, or implicitly by supplying a reverb bounds volume (from which the delay time can be calculated). Well-defined values are floats > 0.0. A value of 0.0 results in reverberation being added as soon as possible after the sound begins. Reverb delay, as calculated from non- null reverb bounds, takes precedence over explicitly set delay time.

Reverberation Bounds

The reverb bounding region defines the overall size of space that reverberation is calculated for. This optional bounds does not have to be the same as the application region of the Soundscape node referencing this AuralAttributes object. If this bounding region is specified then reverb decay and delay are internally calculated from this bounds.

Reverberation Order

The reverb order is a hint that can be used during reverberation to limit the number of late reflections required in calculation of reverb decay. All positive values can be interpreted during reverb rendering as the maximum order of reflections to be calculated. A non-positive value signifies that no limit is placed on the order of reflections calculated during reverberation rendering. In the case where reverb order is not limited, reverb decay is defined strictly by the Reverberation Decay Time parameter.

Decay Time

The reverberation decay time explicitly defines the length of time in milliseconds it takes for the amplitude of late reflections to exponentally decrease to effective zero. In the case where reverb delay is set non-positive the renderer will perform the shortest reverberation decay possible. If ReverbOrder is set, this parameter is clamped by the reverb time calculated as time = reverb Delay * reverb Order. If ReverbOrder is 0, the decay time parameter is not clamped.

Decay Filter

The reverberation decay filter defines how frequencies above a given value are attenuated by the listening space. This allows for modelling materials on surfaces that absorb high frequencies at a faster rate than low frequencies.

Reverberation Diffusion

The reverberation diffusion explicitly defines echo dispersement (sometimes refered to as echo density). The value for diffusion is proportional to the number of echos per second heard in late reverberation, especially noticable at the tail of the reverberation decay. The greater the diffusion the more 'natural' the reverberation decay sounds. Reducing diffusion makes the decay sound hollow as produced in a small highly reflecive space (such as a bathroom).

Reverberation Density

The reverberation density explicitly defines modal reverb density The value for this modal density is proportional to the number of resonances heard in late reverberation perceived as spectral coloration. The greater the density, the smoother, less grainy the later reverberation decay.

Distance Filter

This parameter specifies a (distance, filter) attenuation pairs array. If this is not set, no distance filtering is performed (equivalent to using a distance filter of Sound.NO_FILTER for all distances). Currently, this filter is a low-pass cutoff frequency. This array of pairs defines a piece-wise linear slope for range of values. This attenuation array is similar to the PointSound node's distanceAttenuation pair array, except paired with distances in this list are frequency values. Using these pairs, distance-based low-pass frequency filtering can be applied during sound rendering. Distances, specified in the local coordinate system in meters, must be > 0. Frequencies (in Hz) must be > 0.

If the distance from the listener to the sound source is less than the first distance in the array, the first filter is applied to the sound source. This creates a spherical region around the listener within which a sound is uniformly attenuated by the first filter in the array. If the distance from the listener to the sound source is greater than the last distance in the array, the last filter is applied to the sound source.

Distance elements in these array of pairs is a monotonically-increasing set of floating point numbers measured from the location of the sound source. FrequencyCutoff elements in this list of pairs can be any positive float. While for most applications this list of values will usually be monotonically-decreasing, they do not have to be.

The getDistanceFilterLength method returns the length of the distance filter arrays. Arrays passed into getDistanceFilter methods should all be at least this size.

Doppler Effect Model

Doppler effect can be used to create a greater sense of movement of sound sources, and can help reduce front-back localization errors. The frequency of sound waves emanating from the source are raised or lowered based on the speed of the source in relation to the listener, and several AuralAttribute parameters.

The FrequencyScaleFactor can be used to increase or reduce the change of frequency associated with normal Doppler calculation, or to shift the pitch of the sound directly if Doppler effect is disabled. Values must be > zero for sounds to be heard. If the value is zero, sounds affected by this AuralAttribute object are paused.

To simulate Doppler effect, the relative velocity (change in distance in the local coordinate system between the sound source and the listener over time, in meters per second) is calculated. This calculated velocity is multipled by the given VelocityScaleFactor. Values must be >= zero. If is a scale factor value of zero is given, Doppler effect is not calculated or applied to sound.