Abstract: The present invention provides a method that comprises accepting a plurality of positional data with corresponding times of the motion, determining one or more continuous positional functions that together represent an approximation of path of the positional data, and determining, for each positional function, one or more time functions that together represent an approximation of the times for the positional data corresponding to the positional functions. The present invention provides an apparatus that comprises an input device and a computer system configured to accept the motion signals generated by the input device, determine one or more continuous positional functions that together represent an approximation of the path of the positional data, and determine, for each positional function, one or more time functions that together represent an approximation of the times for the positional data corresponding to the positional functions.

Abstract: In one implementation, a method of transforming a sound into a virtual sound for a synthesized reality (SR) setting is performed by a head-mounted device (HMD) including one or more processors, non-transitory memory, a microphone, a speaker, and a display. The method includes displaying, on the display, an image representation of a synthesized reality (SR) setting including a plurality of surfaces associated with an acoustic reverberation property of the SR setting. The method includes recording, via the microphone, a real sound produced in a physical setting. The method further includes generating, using the one or more processors, a virtual sound by transforming the real sound based on the acoustic reverberation property of the SR setting. The method further includes playing, via the speaker, the virtual sound.

Abstract: A sound signal downmix method includes an inter-channel relationship information obtaining step of obtaining an inter-channel correlation value and an inter-channel time difference in an approximate manner, and a downmix step of obtaining a downmix signal based on the obtained information. In the inter-channel relationship information obtaining step, multiple channel signals are sorted such that signals of adjacent channels are similar to each other, the inter-channel correlation value and the inter-channel time difference are determined only between adjacent channels after the sorting, the inter-channel correlation value between non-adjacent channels is obtained by determining a value that has a monotonically non-decreasing relationship with the inter-channel correlation between the adjacent channels, and the inter-channel time difference between non-adjacent channels is obtained by adding up the inter-channel time differences of adjacent channels.

Abstract: One or more electronic devices include one or more processors that process different segments or channels of an audio signal of sound for a user. The one or more processors process a first segment or channel of the sound with first HRTFs so the first segment or channel externally localizes in front of and to a right of a head of the user and process a second segment or channel of the sound with second HRTFs so the second segment or channel externally localizes in front of and to a left of the head of the user.

Abstract: The inventive method involves receiving as input a representation of an ordered set of two-dimensional images. The ordered set of two-dimensional images is analyzed to determine at least one first view of an object in at least two dimensions and at least one motion vector. The next step is analyzing the combination of the first view of the object in at least two dimensions, the motion vector, and the ordered set of two-dimensional images to determine at least a second view of the object; generating a three-dimensional representation of the ordered set of two-dimensional images on the basis of at least the first view of the object and the second view of the object. Finally, the method involves providing indicia of the three-dimensional representation as an output.

Abstract: An audio engine is provided for acoustically rendering a three-dimensional virtual environment. The audio engine uses geometric volumes to represent sound sources and any sound occluders. A volumetric response is generated based on sound projected from a volumetric sound source to a listener, taking into consideration any volumetric occluders in-between. The audio engine also provides for modification of a level of detail of sound over time based on distance between a listener and a sound source. Other aspects are also described and claimed.

Abstract: The present invention relates to a device and a computerized process for creating a virtual surround sound process for loudspeakers systems, which can create an immersive sound experience with a minimal number of loudspeakers from a less immersive sound source. This computerized process is embodied by software that can be used to process a traditional mono, stereo, or multi-channel audio input to create additional audio channels which are then processed by virtualization filters, and finally summed together with the original input source and output to a loudspeaker system, where a listener will experience a more immersive virtual surround sound experience when compared to listening to only the original source. It should be further noted that the software can be customized for each loudspeaker system for optimal calibration and sonic performance.

Abstract: The transfer function estimation device includes: a correlation matrix computing unit 43 computing a correlation matrix of N frequency domain signals y(f,l); a signal space basis vector computing unit 44 obtaining M vectors v1(f), . . . , vM(f) from eigenvectors of the correlation matrix from highest in the order of corresponding eigenvalues; and a plural RTF estimation unit 45 determining ti(f), . . . , tM(f) that satisfy the relationship of Expression (1), determining a matrix D(f) that is not a zero matrix and that makes ui(f), . . . , uM(f) defined by Expression (2) sparse in a time direction, determining ci,1(f), . . . , cM,N(f) that satisfy the relationship of Expression (3), and outputting c1(f)/c1,j(f), . . . , cM(f)/cM,j(f) as a relative transfer function, where j is an integer of 1 or more and not more than N.

Abstract: Methods, systems, and non-transitory computer readable storage media are disclosed for rendering scene-aware audio based on acoustic properties of a user environment. For example, the disclosed system can use neural networks to analyze an audio recording to predict environment equalizations and reverberation decay times of the user environment without using a captured impulse response of the user environment. Additionally, the disclosed system can use the predicted reverberation decay times with an audio simulation of the user environment to optimize material parameters for the user environment. The disclosed system can then generate an audio sample that includes scene-aware acoustic properties based on the predicted environment equalizations, material parameters, and an environment geometry of the user environment.

Abstract: An audio signal processing method includes generating an initial reflected sound control signal according to a geometrical shape of a virtual space, generating a reverberant sound control signal using a reflected sound parameter of the virtual space, calculating timing of connection, the timing of connection being a point in time at which a volume of an initial reflected sound reproducible by the initial reflected sound control signal is a same as a volume of a reverberant sound reproducible by the reverberant sound control signal, based on the geometrical shape of the virtual space, and increasing, in a period of time before the timing of connection, a level of the reverberant sound control signal so as to cause the volume of the reverberant sound to be closer to the volume of the reverberant sound at the timing of connection.

Abstract: A scene-based beam generation method for ground-to-air coverage based on convex polygon subdivision is provided and includes: obtaining a collection of base station positions by using a method of universal transverse mercator projection; constructing a three-dimensional (3D) low altitude signal coverage area; performing convex polygon subdivision on the area to be covered; and generating a beam configuration of each base station. The method realizes generation of beam configurations of base stations in the 3D low altitude signal coverage area, overcomes a problem that the existing 17 kind of scene-based beams cannot realize the coverage of 3D area, overcomes a problem of mismatch between network state information and beam configuration caused by dynamic adjustment of beam configurations. The beam configurations generated by the method does not need to obtain the number of users in real time to adjust the beam, and has a good coverage ability.

Abstract: A signal processing system and method for delivering spatialized sound, comprising: a spatial mapping sensor, configured to map an environment, to determine at least a position of at least one listener and at least one object; a signal processor configured to: transform a received audio program according to a spatialization model comprising parameters defining a head-related transfer function, and an acoustic interaction of the object, to form spatialized audio; generate an array of audio transducer signals for an audio transducer array representing the spatialized audio; and a network port configured to communicate physical state information for the at least one listener through digital packet communication network.

Abstract: In one implementation, a method of transforming a sound into a virtual sound for a synthesized reality (SR) setting is performed by a head-mounted device (HMD) including one or more processors, non-transitory memory, a microphone, a speaker, and a display. The method includes displaying, on the display, an image representation of a synthesized reality (SR) setting including a plurality of surfaces associated with an acoustic reverberation property of the SR setting. The method includes recording, via the microphone, a real sound produced in a physical setting. The method further includes generating, using the one or more processors, a virtual sound by transforming the real sound based on the acoustic reverberation property of the SR setting. The method further includes playing, via the speaker, the virtual sound.

Abstract: Methods, systems, and program products for generating a virtual studio are disclosed. In one embodiment a method includes processing image information for at least one pinna of a user to generate a head-related transfer function (HRTF) profile of the user. A studio model that includes a studio-specific acoustic profile is accessed such as by a virtual studio client application executing on a laptop. An audio configuration of the studio model is selected based on the studio-specific acoustic profile. An audio media source is activated and the audio configuration is applied in combination with the HRTF profile of the user to audio generated by the audio media source.

Abstract: Systems and methods for limiting volume in an audio playback device using a feedback controller are disclosed herein. In one example, a gain stage modulates gain of an audio signal based in part on feedback from a downstream limiter. The gain stage receives a first audio signal as well as a feedback signal from the feedback controller. Based at least in part on the feedback signal from the feedback controller, the gain stage modulates a gain of the first audio signal to provide a second audio signal. The second audio signal is delivered to the limiter, which limits the second audio signal to produce an output signal. The output signal is played back via a transducer. The feedback controller receives a gain reduction value from the limiter and determines a feedback signal to provide to the gain stage upstream of the limiter.

Abstract: An effect addition device includes at least one processor that executes a time domain convolution process of convolving a first time domain data part of impulse response of sound effects with a time domain data on an original sound, a frequency domain convolution process of convoluting a second time domain data part of the impulse response data with the time domain data on the original sound, a convolution extension process of extending a convolved state(s) of an output signal(s) resulting from the time domain convolution process and/or the frequency domain convolution process by arithmetic processing which corresponds to an all-pass filter and/or arithmetic processing which corresponds to a comb filter, and a synthesized sound effect addition process of adding a sound effect which is synthesized by execution of the time domain convolution process, the frequency domain convolution process and the convolution extension process to the original sound.

Abstract: The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor.

Abstract: A method and apparatus for simulating reverberation in rendering systems. A decoder/renderer may determine how to simulate reverberation with reference to audio scene description format file information or other bitstream information regarding the virtual scene geometry, and/or a lookup table. Late reverberation may be generated based on reverberator parameters derived from the virtual scene geometry, including the lengths of delay lines, attenuation filter coefficients, and diffuse-to-direct ratio characteristics, using a feedback delay network. Early reverberation may be generated based on determining what potential reflecting elements in a virtual scene geometry are reflecting planes; this may involve the simulation of ray reflection, potentially through beam tracing.

Abstract: Methods, systems, and computer program products for rending an audio object having an apparent size are disclosed. An audio processing system receives audio panning data including a first grid mapping first virtual sound sources in a space and speaker positions to speaker gains. The first grid specifies first speaker gains of the first virtual sound sources in the space. The audio processing system determines a second grid of second virtual sound sources in the space, including mapping the first virtual sound sources into the second virtual sound sources of the second virtual sources. The audio processing system selects at least one of the first grid or second grid for rendering an audio object based on an apparent size of the audio object. The audio processing system renders the audio object based on the selected grid or grids.

Abstract: Techniques for regenerating reverberation include generating a reverberation signal based on a source signal and a response of a listening space, generating a compressed reverberation signal by increasing output levels of a first portion of the reverberation signal having output levels below a threshold level more than output levels of a second portion of the reverberation signal having output levels above the threshold level, and combining the compressed reverberation signal with the source signal to form an output signal for the listening space.

Abstract: Disclosed herein are systems and methods for generating and presenting virtual audio for mixed reality systems. A method may include determining a collision between a first object and a second object, wherein the first object comprises a first virtual object. A memory storing one or more audio models can be accessed. It can be determined if the one or more audio models stored in the memory comprises an audio model corresponding to the first object. In accordance with a determination that the one or more audio models comprises an audio model corresponding to the first object, an audio signal can be synthesized, wherein the audio signal is based on the collision and the audio model corresponding to the first object, and the audio signal can be presented to a user via a speaker of a head-wearable device.

Abstract: Ray tracing is performed with sound sources and a listener position in a listening environment, to generate impulse responses associated with each of the sound sources. The impulse responses are combined to form a combined impulse response. One or more filters are determined, each corresponding to the sound sources, based on the combined impulse response and the impulse responses. Each filter serves as a correction factor that holds unique acoustic information for the sound source that the filter is associated with. The combined impulse response and the filters can be applied to one or more audio signals that contain the sound sources, resulting in audio having reverberation that is tailored to the various sound sources. Other aspects are described and claimed.

Abstract: A mixed reality system for displaying virtual holograms using a head mounted display and a tracking device. The HMD has one or more cameras mounted on its exterior connected to a portable computing device. The cameras record what a user's eyes would see in the environment and project the images onto the screen panel display. The cameras also transmit tracking device data to a software application installed on a portable computing device. Light emitting diodes serve as tracking points for the detection algorithm. The tracking device can be configured to transmit data to the portable computing device via optical modulation. Less private radio frequency technologies can be employed such as Bluetooth or WiFi to enable one-way or two-way communication. In some embodiments, multiplexing can be implemented to enable identifying and differentiating multiple tracking devices in a single frame.

Abstract: Methods and systems for intuitive spatial audio rendering with improved intelligibility are disclosed. By establishing a virtual association between an audio source and a location in the listener's virtual audio space, a spatial audio rendering system can generate spatial audio signals that create a natural and immersive audio field for a listener. The system can receive the virtual location of the source as a parameter and map the source audio signal to a source-specific multi-channel audio signal. In addition, the spatial audio rendering system can be interactive and dynamically modify the rendering of the spatial audio in response to a user's active control or tracked movement.

Abstract: An example reverberation device configured for use with a musical instrument is disclosed. The example reverberation device comprises: (a) an input amplifier configured to receive an input signal from the musical instrument and output an amplified input signal based on the received input signal; (b) an input transducer connected to the input amplifier and configured to transmit the amplified input signal; (c) a reverberation plate comprising a bended surface, wherein the reverberation plate is connected to the input transducer, and wherein the reverberation plate is configured to output the input signal as an output signal at the bended surface; (d) an output transducer connected to the reverberation plate at the bended surface and configured to transmit the output signal; and (e) an output amplifier configured to receive the transmitted output signal from the output transducer and output an amplified output signal.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: An analysis filter bank corresponding to a plurality of sub-bands, comprising: multiple sub-filters with different center frequencies which perform multiple complex-type first-order infinite impulse response filtering operations on an audio input signal to generate multiple sub-filter signals; a first set of binomial combiners, each of which performs a weighted-sum operation on a first number of the sub-filter signals with a first set of binomial weights to generate one of multiple sub-band signals; a second set of binomial combiners, each of which performs a weighted-sum operation on a second number of the sub-filter signals with a second set of binomial weights to generate one of multiple lower sub-band-edge signals or one of multiple higher sub-band-edge signals; and multiple envelope detection with decimation devices, which perform multiple envelope detection with decimation operations on the sub-band signals, the lower sub-band-edge signals, and the higher sub-band-edge signals to generate multiple fine

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

Abstract: Disclosed are techniques and devices which include a memory configured to store audio data within a first audio zone, or a second audio zone in a layered soundfield. The memory is coupled to one or more processors and the memory is configured to store the audio data in the first audio zone and the second audio data in the layered soundfield. The one or more processors are configured to receive an interaction command to control the audio data in the first audio zone and the second audio zone in the layered soundfield, and generate one or more indicators that the interaction command was received to control the audio data, in the first audio zone or the second audio zone of the layered soundfield.

Abstract: An illustrative method for estimating a direct-to-reverberant ratio of a sound signal is described, wherein the direct-to-reverberant ratio is indicative of a ratio between direct sound received from a sound source and reverberated sound received from reflections in an environment of the sound source. The method includes determining a first energy value of a sound signal for a first time frame; assigning to an onset value of the first time frame a positive value, if the difference of the first energy value of the first time frame and a second energy value of a preceding second time frame is greater than a threshold, and a zero value otherwise; and determining the direct-to-reverberant ratio by providing an onset signal comprising the onset value to a machine learning algorithm, which has been trained to determine the direct-to-reverberant ratio based on the onset signal.

Abstract: In one implementation, a method of transforming a sound into a virtual sound for a synthesized reality (SR) setting is performed by a head-mounted device (HMD) including one or more processors, non-transitory memory, a microphone, a speaker, and a display. The method includes displaying, on the display, an image representation of a synthesized reality (SR) setting including a plurality of surfaces associated with an acoustic reverberation property of the SR setting. The method includes recording, via the microphone, a real sound produced in a physical setting. The method further includes generating, using the one or more processors, a virtual sound by transforming the real sound based on the acoustic reverberation property of the SR setting. The method further includes playing, via the speaker, the virtual sound.

Abstract: A signal processing device includes a power estimating unit that treats the feature quantity of a signal including reverberation as the input; inputs an observation feature quantity corresponding to an observation signal to a neural network which is learnt in such a way that the estimate value of the feature quantity corresponding to the power of the signal having reduced reverberation, from among the input signal, is output; and estimates the estimate value of the feature quantity corresponding to the power of the signal having reduced reverberation and corresponding to the observation signal. Moreover, the signal processing device includes a regression coefficient estimating unit that uses the estimate value of the feature quantity corresponding to the power as obtained as the estimation result by the power estimating unit, and estimates a regression coefficient of the autoregressive process for generating the observation signal.

Abstract: Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

Abstract: Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

Abstract: The present technology relates to an audio output controller, an audio output control method, and a program that can improve sound quality. An audio output controller includes multiple speaker units installed so as to face different directions, outputs measurement sound from at least one speaker unit of the multiple speaker units, and controls a gain of the speaker unit on the basis of a reverberation characteristic when the measurement sound is measured by a microphone in a predetermined position. Measurement sound output from a speaker unit installed in another audio output controller is measured by the microphone. Alternatively, measurement sound output from an installed speaker unit is measured by the microphone. The present technology can be applied to a wireless speakers, for example.

Abstract: A system for enhancing artificial reverberation in a noisy listening space includes a sensor configured to generate a signal indicating a current noise level in a listening space, a loudspeaker configured to output sound in the listening space based on an output signal, and at least one processor.

Abstract: Determination of material acoustic parameters for a headset is presented herein. A value of a material acoustic parameter is initialized. A simulation is performed using the value of the material acoustic parameter and a model. The model includes a three-dimensional representation of a local area occupied by the headset. During the simulation, the value of the material acoustic parameter is dynamically modified until a reverberation time calculated based on the modified value of the material acoustic parameter falls within a threshold value of a target reverberation time. The model is updated with the modified value of the material acoustic parameter. The model is used to determine one or more acoustic parameters. Audio content is rendered based on the one or more acoustic parameters so that the audio content appears originating from an object in the local area.

Abstract: The disclosure describes one or more embodiments of an impulse response system that generates accurate and realistic synthetic impulse responses. For example, given an acoustic impulse response, the impulse response system can generate one or more synthetic impulse responses that modify the direct-to-reverberant ratio (DRR) of the acoustic impulse response. As another example, the impulse response system can generate one or more synthetic impulse responses that modify the reverberation time (e.g., T60) of the acoustic impulse response. Further, utilizing the synthetic impulse responses, the impulse response system can perform a variety of functions to improve a digital audio recording or acoustic measurement or prediction model.

Abstract: A method, computer program product, and computer system for measuring, by a computing device, a plurality of Room Impulse Responses (RIRs) associated with a set of two or more microphones. At least a portion of the RIRs may be augmented. At least the portion of the RIRs may be converted to their respective Relative Transfer Function (RTF) representations. The RTF representations may be applied to training data to generate an acoustic model for automatic speech recognition.

Abstract: According to the present disclosure, a sound output device includes a sound acquisition part that acquires a sound signal generated from an ambient sound, a reverb process part that performs a reverb process on the sound signal, and a sound output part that outputs a sound generated from the sound signal subjected to the reverb process, to a vicinity of an ear of a listener. This configuration allows a listener to hear sound acquired in real time to which desired reverberation is added.

Abstract: A sound processing device includes: a sound source that outputs a common sound signal to a first speaker for a vehicle exterior and a second speaker for a vehicle interior that are mounted on the vehicle; a notification sound controller that outputs a signal corresponding to a notification sound for notifying an approach of the vehicle to the first speaker based on the sound signal output from the sound source; and an acceleration sound controller that outputs a signal corresponding to an acceleration sound of the vehicle to the second speaker based on the sound signal output from the sound source.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output.

Abstract: Systems and methods for providing accurate and independent control of reverberation properties are disclosed. In some embodiments, a system may include a reverberation processing system, a direct processing system, and a combiner. The reverberation processing system can include a reverb initial power (RIP) control system and a reverberator. The RIP control system can include a reverb initial gain (RIG) and a RIP corrector. The RIG can be configured to apply a RIG value to the input signal, and the RIP corrector can be configured to apply a RIP correction factor to the signal from the RIG. The reverberator can be configured to apply reverberation effects to the signal from the RIP control system. In some embodiments, one or more values and/or correction factors can be calculated and applied such that the signal output from a component in the reverberation processing system is normalized to a predetermined value (e.g., unity (1.0)).

Abstract: A sound processing device includes: a combining processor that combines a performance sound and a source sound, based on operation information corresponding to a performance operation on an instrument. The performance sound is obtained by picking up a sound generated by the performance operation on the instrument. The source sound is obtained from a sound source.

Abstract: Utilizing an adaptive multichannel technique to mitigate reverberation present in received audio signals, prior to providing corresponding audio data to one or more additional component(s), such as automatic speech recognition (ASR) components. Implementations disclosed herein are “adaptive”, in that they utilize a filter, in the reverberation mitigation, that is online, causal and varies depending on characteristics of the input. Implementations disclosed herein are “multichannel”, in that a corresponding audio signal is received from each of multiple audio transducers (also referred to herein as “microphones”) of a client device, and the multiple audio signals (e.g., frequency domain representations thereof) are utilized in updating of the filter—and dereverberation occurs for audio data corresponding to each of the audio signals (e.g., frequency domain representations thereof) prior to the audio data being provided to ASR component(s) and/or other component(s).

Abstract: Certain example embodiments relate to speech privacy systems and/or associated methods. The techniques described herein disrupt the intelligibility of the perceived speech by, for example, superimposing onto an original speech signal a masking replica of the original speech signal in which portions of it are smeared by a time delay and/or amplitude adjustment, with the time delays and/or amplitude adjustments oscillating over time. In certain example embodiments, smearing of the original signal may be generated in frequency ranges corresponding to formants, consonant sounds, phonemes, and/or other related or non-related information-carrying building blocks of speech. Additionally, or in the alternative, annoying reverberations particular to a room or area in low frequency ranges may be “cut out” of the replica signal, without increasing or substantially increasing perceived loudness.

Abstract: Methods and systems for modifying ambient sound waves passing through a wall are disclosed. The wall has a first side and a second side with a cavity therebetween. A sound producing device is situated between the first side and the second side and is configured to produce interfering sound waves in the cavity such that the sound waves passing through the wall are reduced, masked, or reduced and masked.

Abstract: Provided is a signal processing apparatus that can replicate, in a real space, an environment different from the real space by granting an acoustic characteristic different from that of the real space, to a sound released in the real space. There is provided a signal processing apparatus including a control unit configured to decide a predetermined acoustic characteristic for causing a user to hear a collected ambient sound of the user in a space having a different acoustic characteristic, in accordance with content being reproduced, or an action of a user, and to add the decided acoustic characteristic to the ambient sound.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: In music recording and virtual reality applications, it is desired to control the perceived size of a synthesized acoustic space. In an embodiment, a room size parameter is used to modify characteristics of an artificial reverberator so as to affect a listener's sense of the size of the acoustic space. Properties of the reverberation related to perceived room size such as decay time and equalization are adjusted interactively.