Other

Aureal Semiconductor, Inc.

Leland Stanford Junior University

Seismic Innovations

Kind of Loud Technologies, Llc.

University of Limerick

Universal Audio

Jonathan Stuart Abel

Sean A Coffin

Scott Haines Foster

David P Berners

Eoin F Callery

Yoomi Hur

Agnieszka Roginska

Yoo Hsiu Yeh

Julius O Smith

Kurt James Werner

William L Putnam

Ho June Lee

Elliot Kermit Canfield-Dafilou

James W Chaffee

Stacy Lynn Welsh

Eoin Francis Callery

Kyle Swenson Spatt

The present embodiments relate to audio effect processing, and more particularly to a method for creating reverberation impulse responses from prerecorded or live source materials forms the basis of a family of reverberation effects. In one embodiment, segments of audio are selected and processed to form an evolving sequence of reverberation impulse responses that are applied to the original source material—that is, an audio stream reverberating itself. In another embodiment, impulse responses derived from one audio track are applied to another audio track. In a further embodiment, reverberation impulse responses are formed by summing randomly selected segments of the source audio, and imposing reverberation characteristics, including reverberation time, wet equalization, wet-dry mix, and predelay. By controlling the number and timing of the selected source audio segments, the method produces a collection of impulse responses that represent a trajectory through the source material. In so doing, the evolving impulse responses will have the character of room reverberation while also expressing the changing timbre and dynamics of the source audio.

Method and system for artificial reverberation employing reverberation impulse response synthesis

The present embodiments generally relate to enabling participants in an online gathering with networked audio to use a cancelling auralizer at their respective locations to create a common acoustic space or set of acoustic spaces shared among subgroups of participants. For example, there are a set of network connected nodes, and the nodes can contain speakers and microphones, as well as participants and node mixing-processing blocks. The node mixing-processing blocks generate and manipulate signals for playback over the node loudspeakers and for distribution to and from the network. This processing can include cancellation of loudspeaker signals from the microphone signals and auralization of signals according to control parameters that are developed locally and from the network. A network block can contain network routing and processing functions, including auralization, synthesis, and cancellation of audio signals, synthesis and processing of control parameters, and audio signal and control parameter routing.

Networked audio auralization and feedback cancellation system and method

According certain aspects, embodiments of the invention consider the problem of microseismic event localization from a parameter estimation perspective, and include a method and system for computing and displaying characteristics of event localization errors. According to certain other aspects, embodiments of the invention include techniques for deriving aggregate statistics from a set of event location estimates, including methods for computing and displaying the probability that an event occurred in any given volume, and methods for describing and displaying the smallest volume that contains a specified percentage of the event probability or expected to contain the specified percentage of the events.

Method and system for microseismic event location error analysis and display

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.

Method and system for implementing a modal processor

Preparing a sum of complex exponentials in the form of a modal filter to approximate a given impulse response is considered, and a non-iterative solution is described. In one embodiment, the mode count, and the mode frequencies, dampings, and amplitudes are estimated using the generalized eigenvalues of Hankel matrices of samples of the given impulse response.

Method and system for designing a modal filter for a desired reverberation

In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

Method and system for artificial reverberation using modal decomposition

In general, the present invention relates to a method and apparatus for estimating spatial content of a soundfield at a desired location, including a location that has actual sound content obstructed or distorted. According to certain aspects, the present invention aims at presenting a more natural, spatially accurate sound, for example to a user at the desired location who is wearing a helmet, mimicking the sound a user would experience if they were not wearing any headgear. Modes for enhanced spatial hearing may be applied which would include situation-dependent processing for augmented hearing. According to other aspects, the present invention aims at remotely reproducing the soundfield at a desired location with faithful reproduction of the spatial content of the soundfield.

Method and apparatus for estimating spatial content of soundfield at desired location

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.

A method and system for real-time auralization is described in which room sounds are reverberated and presented over loudspeakers, thereby augmenting the acoustics of the space. Room microphones are used to capture room sound sources, with their outputs processed in a canceler to remove the synthetic reverberation also present in the room. Doing so gives precise control over the auralization while suppressing feedback. It also allows freedom of movement and creates a more natural acoustic environment for performers or participants in music, theater, gaming, home entertainment, and virtual reality applications. Canceler design methods are described, including techniques for handling varying loudspeaker-microphone transfer functions such as would be present in the context of a performance or installation.

System and method for augmenting an acoustic space

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.

Artificial reverberator room size control

The present invention relates to methods and apparatuses for using head waves to greatly improve microseismic event localization accuracy, particularly in the depth dimension, by analyzing them in addition to direct path arrivals whenever they are observed. Embodiments of the invention also include techniques known as multipath analysis.

Microseismic event localization using both direct-path and head-wave arrivals

The invention includes a basic approach to simulate the workings of a ribbon microphone based on measurements at an array of more robust small pressure microphones. Embodiments of the invention take an array of microphone elements, either very small microphones that are placed on either side of a printed circuit board or some other device to understand the sound pressure differences from front to back and use those sound pressure differences to emulate the motion of a ribbon if a ribbon were co-located with the array of microphones. The microphone array detects differential pressure, either by a set of elements that that have figure of eight polar patterns, or by having separate elements front and back.

Ribbon array microphone

In the field of seismic data analysis, the problem of estimating features of a microseismic event waveform, detected at the sensors of a geophone array in the presence of background seismic energy and sensor noise, is considered. Embodiments of the present disclosure provide a method and system for estimating wavefront arrival times and polarizations across the sensors of an array, while simultaneously extracting the associated event sensor waveforms from the seismic background. The resulting waveform estimates are estimated in such a way that estimation errors are expected to be noise-like, and free of any systematic changes in the magnitude or phase of the arriving wavefronts.

Growing community of inventors

Menlo Park, CA, United States of America

Jonathan Stuart Abel