Abstract: Audio content is controllably directed to or from an aircraft occupant, such as the pilot or a passenger using a microphone array of at least two transducers, positioned onboard an aircraft to pick up utterances of the aircraft occupant. An electronically steerable transducer array coupled to an onboard an aircraft audio system, such as an avionics communication system or in-flight entertainment system supplies audio content to the aircraft occupant. A signal processor coupled to the microphone array processes utterances sensed by the transducers using a time of arrival algorithm to determine an utterance originating direction. A controller circuit then supplies a pointing direction control signal to the steerable beamforming speaker array or microphone array based on the utterance originating direction to cause the steerable transducer array to direct the audio content in the direction of the sensed utterance and thereby direct the audio content to or from the aircraft occupant.

Abstract: Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: A method to detect the presence and location of submarines in a complex marine environment by wavelet denoising, wavelet signal enhancement, by autocorrelation and signal source identification a convolutional neural network.

Abstract: Embodiments described herein involve an auxiliary zone contributing audio to a primary zone. In an example implementation, a network media system determines that a first zone in the network media system is playing back a first type of audio content and that a second zone in the network media system is not playing back audio content. While the first zone is playing back the first type of audio content and the second zone is not playing back audio content, the network media system forms a temporarily playback configuration. In the temporary playback configuration, the first zone plays back primary audio content including full frequency range audio content and the second playback device of the second zone plays back auxiliary audio content including low frequency range audio content.

Abstract: A hearing aid is described which incorporates an antenna integrated into the housing that is configured to radiate with linear polarization such that the electric field is perpendicular to the head of a wearer. The described technique results in lower propagation losses from ear to ear and an improvement in ear-to-ear communications using a far-field link (e.g., in the 2.4 GHz band).

Abstract: A device may be configured to determining display properties for virtual content in an environment with a plurality of physical participants by capturing an image of the environment, analyzing the captured image to identify at least one object in the environment, determining a parameter for the identified object, and determining a display property of a digital representation of virtual content based on the determined parameter. Embodiments may include negotiating display properties with other devices to generate coordinated display properties, and rendering the digital representation of the virtual content so that the remote participant appears to be in the same fixed position to all co-located participants and sized consistent with the co-located participants.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore, compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: A method for interpolating a sound field captured by a plurality of N microphones each outputting the encoded sound field in a form including at least one captured pressure and an associated pressure gradient vector. Such a method includes an interpolation of the sound field at an interpolation position outputting an interpolated encoded sound field as a linear combination of the N encoded sound fields each weighted by a corresponding weighting factor. The interpolation includes an estimation of the N weighting factors at least from: the interpolation position; a position of each of the N microphones; the N pressures captured by the N microphones; and an estimated power of the sound field at the interpolation position.

Abstract: A robotic avatar apparatus is provided. The robotic avatar apparatus includes a robotic avatar unit. The robotic avatar unit includes a camera, a processor and a unit signal transceiver. The camera is connected to the processor. The processor is connected to the unit signal transceiver. The robotic avatar apparatus also includes a user interface device. The user interface device includes a user signal transceiver and a display. The robotic avatar unit is designed to connect to the user interface device via a connection between the unit signal transceiver and the user signal transceiver.

Abstract: A sound-producing device for a vehicle includes sound-producing means, an opening provided in a vehicle body bottom part, and a duct connecting the sound-producing means to the opening, and it is possible, by amplifying the sound pressure of a sound generated by the sound-producing means through the duct and transmitting the sound to the outside of the vehicle via the opening of the vehicle body bottom part, to inform a pedestrian that the vehicle is approaching and alert them, and to reduce the sound pressure of the sound-producing means compared with a case in which there is no duct. Moreover, since an inlet and an outlet of the duct are offset from each other in a direction that intersects a vertical axis, it is possible to prevent water or small stones scattered by a tire from passing through the duct and to decrease the stiffness of the duct.

Abstract: Methods and apparatus for spatial audio navigation that may, for example, be implemented by mobile multipurpose devices. A spatial audio navigation system provides navigational information in audio form to direct users to target locations. The system uses directionality of audio played through a binaural audio device to provide navigational cues to the user. A current location, target location, and map information may be input to pathfinding algorithms to determine a real world path between the user's current location and the target location. The system may then use directional audio played through a headset to guide the user on the path from the current location to the target location. The system may implement one or more of several different spatial audio navigation methods to direct a user when following a path using spatial audio-based cues.

Abstract: Scene changes should be done pleasingly and without user input or control. Based on the number of speakers and changes in speakers, either to a different individual or movement by the same speaker, based on the locations of the speakers, and based on the overlap of the current and intended scenes, a decision is made whether to perform a smooth transition or do a cut. It has been determined that the decision on cut versus smooth transition is preferably based on the location of the center of the intended new scene versus the boundaries of the current scene, a cut used if the center is outside the boundaries and a smooth transition if inside. If a smooth transition, an easing function, such as ease in ease out, is performed to change the scene. A preferred value for the smooth transition is to perform the transition over 80 frames.

Abstract: Systems and methods for using a digital overlay screen during a video telecommunication. The digital overlay screen comprises content, such as notes or a script, which a presenter may reference during said video telecommunication. The digital overlay screen may tether to an underlying network conferencing program or a video captured from an image capturing device of a user's device. The digital overlay screen is arranged relative to the image capturing device such that the presenter can review the content therein while simultaneously establishing and maintaining a line of sight with the device. The digital overlay screen may overlay on a video feed of the telecommunications without distorting said feed, and without being transmitted or visible to third party recipients of the presentation.

Abstract: Managing noise during an online conference session includes obtaining audio data from an endpoint participating in an online conference session. The audio data is derived from audio captured at the endpoint that includes musical sounds. The audio data is processed to identify a portion of the audio data in which a decibel level of the musical sounds is stable for a period of time. Non-musical noise present, if any, in the audio data with the musical sounds is identified and the non-musical noise is attenuated from the audio data to generate noise-reduced musical audio data. The noise-reduced musical audio data is transmitted for play out at one or more other endpoints participating in the online conference session.

Abstract: A method includes providing, for each respective audio channel of a plurality of audio channels provided by an operating system of a computing device, a set of successive audio processing stages to apply to the respective audio channel. The method also includes providing, by the operating system, an application programming interface (API) configured to set a plurality of parameters for adjusting the set of successive audio processing stages for each respective audio channel. The method additionally includes receiving, via the API and from an application running on the computing device, one or more values for one or more parameters of the plurality of parameters. The method further includes adjusting, by the operating system, the plurality of audio channels based on the received one or more values for the one or more parameters.

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: Embodiments are disclosed for user posture change detection for triggering re-centering of spatial audio. In an embodiment, a method comprises: obtaining source device motion data from a source device and headset motion data from a headset worn by a user; estimating a gravity vector from one of the source device or headset motion data; splitting the source device and headset motion data into vertical and horizontal planes, the vertical plane in the direction of the estimated gravity vector and the horizontal plane perpendicular to the estimated gravity vector; calculating similarity measures based on the source device motion data and headset motion data in the vertical and horizontal planes over a time window; detecting a posture change event based on the calculated similarity measures; and resetting a head tracker error after the detected user posture change event.

Abstract: A conferencing system is configured, for an interval of time, to receive time-dependent input data from a first user, the time-dependent input data obtained via a capturing device. The conferencing system is configured to receive profile data for the first user, analyze the time-dependent input data and the profile data for the first user using a computer-based model to obtain at least one classifier score for a classifier of a reaction of the first user, and transmit the at least one classifier score for the classifier to a second user.