Abstract: A method to audio matrix encode/decode, which encode and decode audio signals of two or more channels into an audio signal of one or more channel while preserving the direction of a sound image includes extracting pieces of sound image information from audio signals of multi channels, encoding and allocating the extracted sound image information to an inaudible frequency domain except an audible frequency domain, and adding the sound image information allocated to the inaudible frequency domain and matrix-encoded stereo signals of the audible frequency domain.

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: A method of measuring and improving acoustic properties, and a recording medium and a music reproduction apparatus that are suitable for the method. The method includes reproducing and outputting an analysis signal stored in codec file types, recording the output analysis signal in the codec file types, obtaining an impulse response by performing convolution on the recorded signal and an inverse signal of the analysis signal, and correcting frequency properties based on the impulse response. By storing a signal that is supposed to be generated by a signal generation device in codec file types, it is possible to implement the signal in a music reproduction apparatus without a need for the signal generation device. By correcting frequency property, the music reproduction apparatus can improve the performance of an output acoustic signal close to that of an original sound.

Abstract: A signal processing apparatus and method providing a 3-dimensional (3D) sound effect may determine a mask related to an ambience of an input signal, separate the input signal into a primary signal and an ambience signal using the mask, decorrelate the ambience signal, and sum the decorrelated ambience signal and the primary signal, accordingly generating an output signal to which a sound effect is applied.

Abstract: The present invention relates to audio signal processing, and more particularly to methods and apparatuses for emulating and controlling various features of mechanical spring reverberation in a digital audio processing system. According to certain aspects of the invention, such an emulation is performed so as to enhance or alter the characteristics of a digitally stored or processed audio signal in substantially the same manner as a mechanical spring reverberation system. In one example embodiment, the propagation of energy through a mechanical spring is simulated using dispersive waveguides, wherein left-going and right-going waves are separately processed, and the effects of dispersion and attenuation commuted to the waveguide ends. According to additional aspects, many spring reverberators contain spring elements arranged in parallel, with no coupling between springs.

Abstract: Improved techniques for presenting sound effects at a portable media device are disclosed. The sound effects can be output as audio sounds to an internal speaker, an external speaker, or both. In addition, the audio sounds for the sound effects can be output together with other audio sounds pertaining to media assets (e.g., audio tracks being played). In one embodiment, the sound effects can serve to provide auditory feedback to a user of the portable media device. A user interface can facilitate a user's selection of sound effect usages, types or characteristics.

Abstract: An effect adding method, includes: applying different gains to a positive side waveform portion and a negative side waveform portion of an audio signal respectively when absolute values of input levels of the positive side waveform portion and the negative side waveform portion are smaller than a predetermined value; producing a higher range component of the audio signal based on a high range component of the audio signal to which the gain is applied, the higher range component being higher in frequency than the high range component; producing a lower range component of the audio signal based on a low range component of the audio signal to which the gain is applied, the lower range component being lower in the frequency than the low range component; and synthesizing an audio signal having an effect sound by adding the audio signal to which the different gains are applied, the higher range component, and the lower range component with each other.

Abstract: An array speaker apparatus SParray includes a first radiation means for driving speaker units so that a first sound S1 of a main channel is radiated to a wall surface W1 on the left or right side of a listening position, and a second radiation means for driving the speaker units so that a second sound S2 the same as the first audio signal is radiated directly to the listening position.

Abstract: A system and method for wirelessly triggering an amplification system are disclosed. In one example, the system for wirelessly triggering the amplification comprises at least one wireless tag, the at least one wireless tag having unique identification (ID) information and an amplification system configured to receive audio signals. Further, the amplification system comprises a wireless reader configured to receive and determine the ID information from the at least one wireless tag, and a processor configured to associate at least one sound effect with the ID information received from the at least one wireless tag and to modify the audio signals to include the at least one sound effect.

Abstract: Each of level meter block images 21 to 2m arranged on a touch panel display 1 displays, as a bar graph, respective signal levels of channels of a corresponding one of m blocks into which a plurality of channels are divided. The level meter block images 21 to 2m also serve as assignment selection buttons of an assignment selection portion 2. For each level meter block image, an image generation portion 8 generates not only an image of a bar graph but also an image which indicates respective color indicators of the respective channels such that, when a signal type of a channel indicates that there is a signal correlation between the channel and its neighboring channel, the channel has a color indicator of a connected shape shared with the neighboring channel, and synthesizes these images to output the synthesized image to an image display control circuit 3.

Abstract: The present disclosure provides a digital audio signal processing system that comprises a set of delay lines, allpass and lowpass filters to achieve the reverberation effect. The present disclosure further provides a method for generating and controlling digital reverberations for audio signals. The reverberation generated will have an increasing echo density in the time domain and a faster decay of high frequency signals than low frequency signals. The controlling mechanism of reverberation generation is realized through the extraction of the real environment characteristics.

Abstract: A therapeutic method and sound recording including sounding a first harmonic interval for a first duration of at least approximately one minute; and sounding a second harmonic interval for a second, consecutive duration for at least approximately one minute. The first and second harmonic intervals may be just intervals and/or fifths, such as Perfect 5ths (P5). The first duration may overlap with the second duration in order to sound a Major Seventh (M7) chord. The first duration may be approximately twenty minutes long or longer, the second duration may be approximately ten minutes long or longer, and the overlap between the first and second duration may be approximately ten minutes long or longer.

Abstract: A digital audio signal processor uses a matrix transform to produce a multi-channel synthetic reverberation signal based on variably-delayed versions of an input signal. The matrix transform outputs are mixed with the inputs to the signal delaying mechanisms, and some of the outputs are further processed to create the reverberation signal.

Abstract: A speaker virtualization system provides virtual surround sound using a pair of physical loudspeakers. A multiple surround audio channels input is processed using a combination of head related transfer functions and shaped reverberation to provide widening and front/back auditory clues without requiring any kind of interaural path cancellation. The system uses a 360 degree power-response head related transfer function to provide perceptual separation of the reverberant and direct paths, along with discrete, different reverberation for left and right rendering channels to provide envelopment. By eliminating interaural path cancellation, the speaker virtualization system also produces a wider virtual surround sound effect, without dependency on head position and facing.

Abstract: An engine sound generation device adapted to an electric vehicle includes an engine speed calculation unit which calculates the virtual engine speed based on the motor revolution speed, vehicle speed, and acceleration opening. The engine speed calculation unit calculates the virtual engine speed with reference to the vehicle-dependent engine speed memory on the condition that the virtual engine speed is deemed proportional to the vehicle speed. In addition, the engine speed calculation unit calculates the virtual engine speed with reference to the vehicle-nondependent engine speed memory on the condition that the virtual engine speed is not deemed proportional to the vehicle speed, thus generating the common engine sound matching the behavior of an engine in a prescribed event such as the startup of the electric vehicle and the semi-engagement of a clutch.

Abstract: Calculation is performed for sound paths 112-1, 114-1 along which sounds emitted from a sound emitting point 104 in an acoustic space 102 are reflected and delivered to a sound receiving point 106. By the calculation, entering angles eR1, eR2 by which the sound paths enter the front side 106a of the sound receiving point 106 are obtained. Calculation is then performed to obtain angles by which respective speakers 52C, 52L, 52R, 52SR, 52SL of a 5.1 surround system are arranged in a listening room, with the front side 106a of the sound receiving point 106 centered thereon. Audio signals on the respective sound paths are distributed among channels for any two speakers. Consequently, sharp localization of sound images is achieved, requiring less calculation in simulating acoustic characteristics of the acoustic space 102 in which the sound emitting point 104 for emitting sounds and the sound receiving point 106 for receiving the sounds are placed.

Abstract: An engine sound synthesizer includes an engine sound storing section, and a synthetic engine sound data generating section which generates synthetic engine sound data on the basis of engine sound data stored in the engine sound storing section. The engine sound storing section stores therein plural groups of plural entities of engine sound data recorded in different driving states, the plural groups of plural engine sound data entities being stored therein in correspondence with predetermined plural driving state ranges. The synthetic engine sound data generating section generates the synthetic engine sound data by selectively reading out of the engine sound storing section plural engine sound data entities in a group corresponding to one of the driving state ranges specified by externally input driving state specification information and selectively combining the read engine sound data entities. The driving states are each defined, for example, by a throttle opening degree and an engine rotation speed.

Abstract: The invention relates to sonic steerable antennae and their use to achieve a variety of effects. The invention comprises a method and apparatus for taking an input signal, replicating it a number of times and modifying each of the replicas before routing them to respective output transducers such that a desired sound field is created. This sound field may comprise a directed beam, focus beam or a simulated origin. Further, “anti-sound” may be directed so as to create nulls (quiet spots) in an already existing sound field. The input signal replicas may also be modified in way which changes their amplitude or they may be filtered to provide the desired delaying.

Abstract: The present invention is based on the finding that a sound-reproduction system, which can generate a spatial sound impression in a reproduction environment, can be efficiently and intuitively controlled by means of a graphic user interface when an impulse response associated with a spatial direction with respect to the reproduction environment is graphically represented, and when the possibility is created for a user to change the impulse response graphically so that, based on the user input of a change, the changed impulse response can be graphically represented and the changed graphical representation can be detected, in order to control the sound-reproduction system. Since it is system-theoretically possible to describe all known linear signal processing operations by impulse responses, it is possible, with the inventive graphic user interface, to provide a sound mixer, through a graphical representation, with an intuitive access to sound effects depending on the direction.

Abstract: A vehicle engine harmonic enhancement system. A method for operating the system includes providing a first engine harmonic enhancement audio signal and equalizing the first engine harmonic enhancement audio signal separately for each of the plurality of loudspeakers to provide individually equalized loudspeaker engine harmonic enhancement audio signals. A vehicle audio system includes a plurality of loudspeakers, an entertainment audio system, and an engine harmonic enhancement system. A method includes equalizing an entertainment audio signal to provide an equalized entertainment audio signal; providing an engine harmonic enhancement audio signal; equalizing the engine harmonic enhancement audio signal separately from the entertainment audio signal to provide an equalized engine harmonic enhancement audio signal; and combining the equalized entertainment audio signal with the equalized engine harmonic enhancement audio signal.

Abstract: A method and system for performing audio signal processing is provided. A phase of a signal may be shifted between 0° and 360° for all frequencies of the signal, without any significant delay distortion. The phase shift can be accomplished through the use of two signal outputs (which are each shifted in phase by 90° relative to the other), inverters and an analog summing circuit. Selective use of the inverter on one or both of the 90° phase shifted outputs, and their linear combinations, allows the phase of the audio signal outputs to be shifted between about 0° and about 360° by a recording engineer, for example.

Abstract: A recording and/or reproducing apparatus includes a microphone, a semiconductor memory, an operating section and a controller. An output signal from the microphone is written in the semiconductor memory and the written signals are read out from the semiconductor memory. The operating section performs input processing for writing a digital signal outputted by an analog/digital converter, reading out the digital signal stored in the semiconductor memory and for erasing the digital signal stored in the semiconductor memory. The control section controls the writing of the microphone output signal in the semiconductor memory based on an input from the operating section and the readout of the digital signal stored in the semiconductor memory.

Abstract: A system and method to modify audio communications in an online environment and, in particular, to modifying audio communications using one or more parameters. The system includes a module configured to detect one or more parameters applicable to an incoming audio communication and a weighing engine configured to weigh the one or more parameters and obtain a weighted volume. The system further includes a module configured to transmit the weighted volume to a user participating in a virtual universe. The method determines whether one or more parameters apply to an incoming communication, weighs the incoming communication based on the one or more parameters to determine a weighted volume of the incoming communication, and transmits the incoming communication to a user at the weighted volume.

Abstract: The disclosure provides a method for noise generation, including: determining an initial value of a reconstructed parameter; determining a random value range based on the initial value of the reconstructed parameter; taking a value in the random value range randomly as a reconstructed noise parameter; and generating noise by using the reconstructed noise parameter. The disclosure also provides an apparatus for noise generation.

Abstract: Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least on DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP's processing of audio data. From the received impulse response parameter, the application identifies a second audio track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.

Abstract: The invention relates to a method for synthesizing an engine noise, in particular of an internal combustion engine, wherein the engine noise is generated by at least one electromechanical transducer, in particular an actuator or a loudspeaker, by means of a signal value corresponding to an electrical transducer excitation signal.

Abstract: A favorable high-frequency signal is generated and practical high-frequency signal interpolation is implemented through simple processing. A digital audio signal reproduced by an instrument, which also carries out compression, is supplied as an original signal to an input terminal 1, and this original signal is then supplied to a digital sample and hold circuit 3 via a band-pass filter 2. The signal from the digital sample and hold circuit 3 is supplied to a ±1 multiplier 6, which then alternately inverts sign bits. The harmonic components of this signal in which the sign bits are inverted alternately are extracted by a high-pass filter (HPF) 7. Meanwhile, the original signal from the input terminal 1 is supplied to a delay circuit 8 equivalent to the processing time consumed by the aforementioned digital sample and hold circuit 3 and related circuits, forming an adjusted, delayed signal.

Abstract: A sensor is provided in a machine having a continuously variable transmission, for example, and the amount of force exerted by the machine against a load is sensed. Based on the sensed power, an appropriate audio signal is generated which simulates the sounds an operator would expect to hear when such force is applied with a machine having a conventional geared transmission. The sounds can include engine noises, as well as sounds associated with the flow of oil or other hydraulic fluids. In addition, the sounds can be generated from digitally pre-recorded audible data, or can be output based on calculated audible data. User controls are also provided to adjust the volume and frequency or pitch of the generated sounds.

Abstract: A composite sound dampening structure includes a first base layer of sound dampening material extending around and against an inside surface of a container and a second wedge layer of sound dampening material attached to an inside surface of the first base layer. The composite sound dampening structure provides improved acoustic dampening in relative small sound chambers. An audio test system generates a composite audio signal of multiple different audio signals that are combined together using linear superposition. The composite audio signal allows a device to be simultaneously tested with multiple different audio frequencies.

Abstract: A method of modeling a room impulse response according to an embodiment of the invention includes receiving a sound pressure signal that is obtained by a microphone when an impulse-type sound source is excited and detecting a room impulse response; determining boundaries between a plurality of intervals of the room impulse response such that the room impulse response is divided into the plurality of intervals on a time domain; and modeling the room impulse response for each of the plurality of divided intervals using at least two different modeling schemes.

Abstract: A sound effect circuit and processing method are disclosed. An input audio signal is input to a digital sound effect unit, an output signal is generated and output with a sound effect in accordance with an effect amount of a certain value from the input audio signal using the digital sound effect unit, the input audio signal is also input to a silence state detection unit, a silence detect signal is generated and output when detecting that the current state of the input audio signal is a silence state using the silence state detection unit, the output signal is input to a sound effect amount control unit, and an output audio signal is output using the sound effect amount control unit, wherein the sound effect amount control unit changes the effect amount to be smaller than the certain value when the silence detect signal is generated.

Abstract: Improved techniques for presenting sound effects at a portable media device are disclosed. The sound effects can be output as audio sounds to an internal speaker, an external speaker, or both. In addition, the audio sounds for the sound effects can be output together with other audio sounds pertaining to media assets (e.g., audio tracks being played). In one embodiment, the sound effects can serve to provide auditory feedback to a user of the portable media device. A user interface can facilitate a user's selection of sound effect usages, types or characteristics.

Abstract: A method and system for generating and/or performing music in real time can include receiving one or more audio signals, receiving one or more virtual instrument trigger signals, and selecting one or more plug-ins and/or one or more virtual instruments. A processing scheme may be selected from a set of operations. The received audio signals and instrument trigger signals are processed in real time as a function of the selected plug-ins, virtual instruments and processing scheme, and outputted in real time as music signals.

Abstract: There is provided a method of customizing an audio presentation for tailoring to a predetermined event sequence, one embodiment comprising identifying the predetermined event sequence from a predetermined event sequence database, associating a first audio asset with one or more selected events of the predetermined event sequence, synchronizing the timing of the first audio assets with the timing of the selected events, and linking the first audio assets to produce a customized audio presentation tailored to the predetermined event sequence. In one embodiment, the method includes downloading an interactive audio presentation content including an interactive audio presentation program, an audio assets database, and the predetermined event sequence database to a client computer. In one embodiment, a system for customizing an audio presentation tailored to a predetermined event sequence comprises a client computer, a client memory located on the client computer, and interactive audio presentation content.

Abstract: In a user terminal, a reproducing unit reads out from a music file memory unit a music file selected and input by a user and reproduces the music file. In a normal reproduction, an output unit outputs the reproduced sound as an audio. When a notification information reception unit receives information that needs to be noticed such as an e-mail or a phone call, a notification information analysis unit analyzes the setting of the information or notification sound and determines the degree of importance. A frequency band allocation unit selects from allocation patterns of frequency bands stored in an allocation information memory unit a pattern for the music and for the notification sound according to the degree of importance and the frequency band of the notification sound. An audio processing unit extracts and synthesizes the components of the frequency bands allocated to music and the notification sound, and the output unit outputs accordingly.

Abstract: A method and apparatus for processing an audio signal to enhance the perceived lower frequency content of the audio signal when played through an audio output device, includes an input configured to receive an audio input signal, a processor configured to filter the audio input signal to produce a high frequency signal and a low frequency signal, generate an enhancement signal by producing higher frequency harmonics from the low frequency signal, including a process of self convolution, and combine the high frequency signal with the enhancement signal to produce an output signal; and an output configured to receive the output signal and produce an audio output.

Abstract: An audio data generation method comprises generating a first parametric description of features of a first sound. The first parametric description comprises a first set of parameters which relates to the features of the first sound. The method comprises generating a second parametric description of features of a second sound. The second parametric description comprises a second set of parameters which relates to the features of the second sound. The to method further comprises generating audio data for output based on a combination of one or more properties of the first parametric description and one or more properties of the second parametric description.

Abstract: An electrical signal derived from an audio source is passed through an amplifier and is then divided into two discrete signal paths. The signal traveling along the first path travels through a high pass filter and is then split so that a portion is delayed and a portion passes through a harmonics generator so that the harmonics are advanced ahead of the delayed portion from the high pass filter. The split circuit is then combined in a summing amplifier. The second portion of the signal is also delayed before it also travels to the mixer. The combined signal is passed through an amplifier and then fed into an output transducer. The use of a delay both on the input signal and the portion of the signal which passed through the high pass filter output just before the summing amplifier enables the harmonics to not be masked by the lower frequencies, making the effect more apparent.

Abstract: According to one embodiment, a sound corrector includes a signal outputter, a response signal, a frequency specifier, a coefficient specifier, a filter, and an outputter. The signal outputter outputs a measurement signal to measure acoustical properties of an object to be measured. The response signal receiver receives a response signal from the object in response to the measurement signal. The frequency specifier specifies a resonant frequency at a resonance peak from the response signal. The coefficient specifier specifies a correction coefficient of a correction filter for reducing the resonant frequency based on the specified resonant frequency. The filter performs filtering on a signal to be output to the object using the correction filter with the correction coefficient. The outputter outputs the signal having undergone the filtering to the object.

Abstract: A delay time calculation apparatus that enables all of speaker units constituting a speaker array to contribute to the formation of a combined wavefront directed to an area specified by a user. The delay time calculation apparatus includes a delay time calculation unit that calculates delay times of delayed audio signals supplied to the speaker units such that a ratio at which an evaluation object area is occupied by an area to which an acoustic wave output from each speaker unit reaches earlier than acoustic waves output from the other speaker units falls within a predetermined range. The evaluation object area is a target area to which a combined wavefront of acoustic waves output from the speaker units is directed, or is a perspective projection image of the target area onto a predetermined evaluation plane.

Abstract: Bass frequencies of audio can be dynamically boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. Bass frequencies of audio can be boosted using a linear combination of an input audio signal and output of a high-pass filter. For example, bass frequencies of audio can be boosted by applying a high-pass filter to an input audio signal to produce an output of the high-pass filter, determining a current level, determining a target gain amount, dynamically adjusting the input audio signal and the output of the high-pass filter, and combining the gain-adjusted signals to produce an output signal. A dynamic bass boost filter can comprise a high-pass filter and a dynamic boost module.

Abstract: A method for digitally processing audio signals to emulate the effects of vacuum tube amplifiers and preamplifiers, musical instrument amplification systems, and distortion effects. By use of an implicit numerical method to estimate the response of a parametrically-controlled non-linear transfer function, non-linear filters, and feedback elements, the dynamic behavior and distortion effects of tube amplification stages are simulated. This provides the capability to reproduce the desired sounds of vintage and modern tube amplifier systems and effects with the conveniences and control associated with digital signal processing systems and software.

Abstract: A system and methods for distortion reduction in a non-linear audio system are disclosed. A method includes the operation of defining a non-linear system transfer function which relates a system electronic input signal with a system acoustic output over a range in which the system acoustic output is in a transition between non-saturated and saturated conditions in a medium in which the system acoustic output of the non-linear audio system is broadcast. A preprocessing transfer function is created that is the inverse of the non-linear system transfer function. The preprocessing transfer function is applied to the system electronic input signal to reduce the level of harmonic distortion at the system acoustic output.

Abstract: A method of acoustically rendering a virtual environment is described. The method includes receiving a subset of polygons derived for an acoustic display from a set of polygons generated for a graphical display. Acoustic reflections are determined from a sound source that bounce off of polygons in the subset of polygons to a listener position in the virtual environment. It is determined whether a polygon in the subset of polygons causes an occlusion of the sound source at the listener position, and a play list of sounds is generated based on the reflections and occlusions.

Abstract: A method for digitally processing audio signals to emulate the effects of vacuum tube amplifiers and preamplifiers, musical instrument amplification systems, and distortion effects. By use of a parametrically-controlled non-linear transfer function, non-linear filters, feedback elements, and power-law function models, the dynamic behavior and distortion effects of tube amplification stages are simulated. This provides users with the capability to reproduce the desired sounds of vintage and modern tube amplifier systems and effects with the conveniences and control associated with digital signal processing systems and software.

Abstract: An acoustic processing device, includes an inputting section which receives a first audio signal and a second audio signal, a first input signal amplifying section which amplifies the first audio signal at an amplification factor, a second input signal amplifying section which amplifies the second audio signal at an amplification factor, a first initial delayed amplifying section, a first adding section which adds the first audio signal amplified by the first amplifying portion and the first audio signal amplified by the first input signal amplifying section, a second adding section which adds the first audio signal amplified by the second amplifying portion and the second audio signal amplified by the second input signal amplifying section, a first supplying section which supplies an audio signal output from the first adding section to a first sound emitting section, and a second supplying section which supplies an audio signal output from the second adding section to a second sound emitting section differen

Abstract: Audio signal processing apparatus is disclosed. The audio signal processing apparatus includes a first audio signal extracting section, a second audio signal extracting section, a sense-of-depth controlling section, a sense-of-sound-expansion controlling section, a control signal generating section, and a mixing section. The first audio signal extracting section extracts a main audio signal. The second audio signal extracting section extracts a sub audio signal. The sense-of-depth controlling section processes the extracted main audio signal to control a sense of depth. The sense-of-sound-expansion controlling section processes the extracted sub audio signal to vary a sense of sound expansion. The control signal generating section generates a first control signal with which the sense-of-depth controlling section is controlled and a second control signal with which the sense-of-sound-expansion controlling section is controlled.

Abstract: An engine sound control apparatus capable of arranging a sound of a frequency that is lower than the fundamental-order frequency of the engine. The engine sound control apparatus detects the frequency of the engine sound and generates a sound signal of a frequency that is in accordance with a control signal and outputting the sound signal generated by the oscillator. Through the interaction between the sound generated by the apparatus and the engine sound, occupants in the passenger compartment will perceive a sound of a frequency that is equal to the difference between the generated sound and the fundamental-order sound. This yields a sound that is lower in frequency than the fundamental-order sound without actually generating a sound of a frequency that is lower than that of the fundamental-order sound.

Abstract: Reverberation parameters for one or more positions of interest are derived from graphics data used for displaying a computer-generated environment. For each position of interest for which reverberation parameters are desired, environmental parameters including distances and the hardness of features in a range of interest and at points on cubemap faces are automatically determined from the graphics data. The environmental parameters are stored with the graphics data and associated with each position of interest. Upon rendering of the computer-generated environment, reverberation property set values usable by a reverberation engine are calculated or interpolated between predetermined values according to the environmental parameters.

Abstract: Embodiments of the invention provide method, system, and computer readable storage medium for providing an audio motion feedback to a user of a simulated three-dimensional (“3D”) environment, where the user controls an entity within the simulated 3D environment. A plurality of gridlines is projected over the simulated 3D environment to form a virtual 3D grid. A crossing of at least one boundary point by the entity is detected, wherein the boundary point is located on or within a threshold distance from at least one of the plurality of the projected gridlines. At least one boundary point type of the crossed boundary point is determined. At least one audio motion signal associated with the determined boundary point type is generated, thus providing audio motion feedback to the user regarding the entity's movements in the simulated 3D environment.