Dolby Laboratories Licensing Corporation

Sonical Sound Solutions

Gary Spittle

Glenn N Dickins

Xuejing Sun

Michael P Hollier

Kai Lin Li

David Gunawan

Michael Eckert

Paul Andrew Boustead

David S McGrath

Thao D Hovanky

Kai Li

Doh-Suk Kim

David D Clementson

Paul Francis Holmberg

Guilin Ma

Gopal Erinjippurath

Erwin Goesnar

Venugopal Kalluri

Craig Johnston

Jean-Francois Catz

Anthony Tucker

Vinh Nguyen

David Clementson

David S Mcgrath

Paul Holmberg

Disclosed herein is an audio system that can be customized by the user (including the developer) and that may allow the user to control more than just the sound levels and may allow the user to select more than one of a few pre-determined settings. Disclosed herein is a development platform that allows manufacturers (including the developers) to test various different parameters, settings, algorithms, functions, processes, accessories, and the like. The development platform allows the manufacturer to simulate the performances from possible options for the audio system, and then load the selected option from the development platform to the audio system. Additionally, the development platform that enables the end user to have the same level of control and ability to access, upload, and control plugins on the end user's device(s).

Fully customizable ear worn devices and associated development platform

Some implementations involve analyzing audio packets received during a time interval that corresponds with a conversation analysis segment to determine network jitter dynamics data and conversational interactivity data. The network jitter dynamics data may provide an indication of jitter in a network that relays the audio data packets. The conversational interactivity data may provide an indication of interactivity between participants of a conversation represented by the audio data. A jitter buffer size may be controlled according to the network jitter dynamics data and the conversational interactivity data. The time interval may include a plurality of talkspurts.

Jitter buffer control based on monitoring of delay jitter and conversational dynamics

An audio apparatus is configured to switch, when there exists a first audio interface between the audio apparatus and a computer apparatus, to using a second audio interface between the audio apparatus and the computer apparatus, the second audio interface being different from the first audio interface. The switching comprises: receiving, via the first audio interface, combined audio data and non-audio data, the non-audio data comprising a request to switch to using the second audio interface; obtaining the request from the data; and, in response to obtaining the request, transmitting to the computer apparatus a confirmation of switching to using the second audio interface. The audio apparatus and the computer apparatus are described and claimed.

Switching to a second audio interface between a computer apparatus and an audio apparatus

In an audio conferencing environment, including multiple users participating by means of a series of associated audio input devices for the provision of audio input, and a series of audio output devices for the output of audio output streams to the multiple users, with the audio input and output devices being interconnected to a mixing control server for the control and mixing of the audio inputs from each audio input devices to present a series of audio streams to the audio output devices, a method of reducing the effects of cross talk pickup of at least a first audio conversation by multiple audio input devices, the method including the steps of: (a) monitoring the series of audio input devices for the presence of a duplicate audio conversation input from at least two input audio sources in an audio output stream; and (b) where a duplicate audio conversation input is detected, suppressing the presence of the duplicate audio conversation input in the audio output stream.

Nearby talker obscuring, duplicate dialogue amelioration and automatic muting of acoustically proximate participants

A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.

Audio capture and render device having a visual display and user interface for use for audio conferencing

A method of outputting audio in a teleconferencing environment includes receiving audio streams, processing the audio streams according to information regarding effective spatial positions, and outputting, by at least three speakers arranged in more than one dimension, the audio streams having been processed. The information regarding the plurality of effective spatial positions corresponds to a perceived spatial scene that extends beyond the speakers in at least two dimensions. In this manner, participants in the teleconference perceive the audio from the remote participants as originating at different positions in the teleconference room.

System and method of speaker cluster design and rendering

The present document relates to methods and systems for setting up and managing two-dimensional or three-dimensional scenes for audio conferences. A conference controller () configured to place a plurality of upstream audio signals () associated with a plurality of conference participants within a 2D or 3D conference scene to be rendered to a listener () is described. The conference controller () is configured to set up a X-point conference scene with X different spatial talker locations () within the conference scene; assign the plurality of upstream audio signals () to respective ones of the talker locations (); determine a degree of activity of the plurality of upstream audio signals (); determine a dominant one of the plurality of upstream audio signals (); and emphasize the dominant upstream audio signal ().

Schemes for emphasizing talkers in a 2D or 3D conference scene

An audio spatial rendering apparatus and method are disclosed. In one embodiment, The audio spatial rendering apparatus includes a rendering unit for spatially rendering an audio stream so that the reproduced far-end sound is perceived by a listener as originating from at least one virtual spatial position, a real position obtaining unit for obtaining a real spatial position of a real sound source, a comparator for comparing the real spatial position with the at least one virtual spatial position; and an adjusting unit for, where the real spatial position is within a predetermined range around at least one virtual spatial position, or vice versa, adjusting the parameters of the rendering unit so that the at least one virtual spatial position is changed.

Audio spatial rendering apparatus and method

A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved.

Matching reverberation in teleconferencing environments

The present document relates to setting up and managing two-dimensional or three-dimensional scenes for audio conferences. A conference controller () configured to place an upstream audio signal () associated with a conference participant within a 2D or 3D conference scene to be rendered to a listener () is described. An X-point conference scene with X different spatial talker locations () is set up within the conference scene, wherein the X talker locations () are positioned within a cone around a midline () in front of a head of the listener (). A generatrix () of the cone and the midline () form an angle which is smaller than or equal to a pre-determined maximum cone angle. The upstream audio signal () is assigned to one of the talker locations () and metadata identifying the assigned talker location () are generated, thus enabling a spatialized audio signal.

Placement of talkers in 2D or 3D conference scene

A conference controller () configured to place an upstream audio signal () associated with a conference participant and a sound signal within a 2D or 3D conference scene to be rendered to a listener () is described. The conference controller () is configured to set up a X-point conference scene with X different spatial talker locations () within the conference scene, X being an integer, X&#x3e;0; assign the upstream audio signal () to one of the talker locations (); place a sound signal at a spatial sound location () within the X-point conference scene; and generate metadata identifying the assigned talker location () and the spatial sound location and enabling an audio processing unit () to generate a spatialized audio signal based on a set of downstream audio signals () comprising the upstream audio signal () and the sound signal.

Placement of sound signals in a 2D or 3D audio conference

A telecommunications device () comprises a base part (), a cap part (), a body part (), a control panel (), an upper light ring (), a lower light ring (), a plurality of sound microphones and a plurality of sound speakers. The cap part () is concave in shape. The body part () is frusto-conical in shape and tapers outwardly away from the cap part () towards the base part (). The body part () and the cap part () cover the sound microphones and the sound speakers. The control panel () is inclined relative to the plane of the support surface () and relative to the plane of the base part (). The control panel () protrudes upwardly from the upper edge of the body part () over the concave cap part (), and the control panel () protrudes downwardly from the lower edge of the body part (). The control panel () controls operation of the sound microphones and the sound speakers. The upper light ring () is located between the cap part () and the body part (). The lower light ring () is located between the body part () and the base part (). When the device () is in an active state in which the device () is capable of capturing sound and/or rendering sound, each of the light rings () emits light to indicate the active state of the device ().

Telecommunications device

From a plurality of received voice signals, a signal interval in which there is a talker collision between at least a first and a second voice signal is detected. A processor receives a positive detection result and processes, in response to this, at least one of the voice signals with the aim of making it perceptually distinguishable. A mixer mixes the voice signals to supply an output signal, wherein the processed signal(s) replaces the corresponding received signals. In example embodiments, signal content is shifted away from the talker collision in frequency or in time. The invention may be useful in a conferencing system.

Growing community of inventors

Hillsborough, CA, United States of America

Gary Spittle