Hewlett-packard Development Company, L.P.

Bowon Lee

Kar-Han Tan

Ton Kalker

Ronald W Schafer

Debargha Mukherjee

Ian N Robinson

Amir Said

Souvik Sen

Victor C Lee

Majid Fozunbal

Kar Han Tan

Thomas Anthony Baran

Example implementations relate to determining a location using time difference of arrival (TDOA). For example, a computing device may include a first sensor to receive a signal at a first time, where the signal is generated by a user contact at a particular location on a keyboard associated with the computing device. The computing device also includes a second sensor to receive the signal at a second time and a third sensor to receive the signal at a third time. The computing device may also include a processor. The processor may calculate a set of TDOAs associated with the first time, the second time, and the third time. The processor may determine the particular location of the user contact using a triangulation based on the set of TDOAs and may identify a character on the keyboard, where the character is associated with the particular location.

Determining location using time difference of arrival

An example provides an apparatus to receive an audio input signal, generate, from the audio input signal, a filtered audio input signal to suppress at least one modal frequency of a flat loudspeaker diaphragm, and output the filtered audio input signal to at least one actuator to cause a vibration of the flat loudspeaker diaphragm.

Suppressing a modal frequency of a loudspeaker

An audio processor for a video conference system receives an audio signal from content to be shared over a video conference and an audio signal from a network. The audio signal from the shared content and the audio signal from the network are mixed together for output to a speaker. The audio processor may also receive a local audio signal from a microphone. The local audio signal is mixed with the audio signal of the shared to content to generate an outbound signal.

Audio processing of shared content

Examples disclose a method to receive a first audio signal at a first microphone positioned at an actual distance from a second microphone. Additionally, the examples disclose the method is further to receive a second audio signal at the second microphone, the second audio signal is associated with an actual time delay relative to the first audio signal. Also, the examples disclose the method is also to determine a virtual time delay corresponding to a virtual distance that is different from the actual distance and to obtain a spatial audio signal based the distances and the time delays.

Obtaining a spatial audio signal based on microphone distances and time delays

The present invention describes a method of determining the active talker for display on a video conferencing system, including the steps of: for each participant, capturing audio data using an audio capture sensor and video data using a video capture sensor; determining the probability of active speech (p, p. . . p), where the probability of active speech is a function of the probability of soft voice detection captured by the audio capture sensor and the probability of lip motion detection captured by the video capture sensor; and automatically displaying at least the participant that has the highest probability of active speech.

System and method for determining the active talkers in a video conference

Techniques for multi-user videoconferencing are described herein. The techniques may include receiving two or more audiovisual streams from a remote site, wherein the two or more audiovisual streams are associated with video images to be rendered at a display of a local computing device. The techniques may include rendering the video images associated with the audiovisual streams at the local display in an arrangement such that a part of one of the video images associated with the audiovisual streams is relatively more opaque than a part of a video image associated with the other audiovisual stream, wherein the arrangement of video images associated with the audiovisual streams is based on cues, the cues comprising video signals indicating activity existing within an audiovisual stream.

Arrangement of multiple audiovisual streams

A method for time delay estimation performed by a physical computing system includes passing a first input signal obtained by a first sensor through a filter bank to form a first set of sub-band output signals, passing a second input signal obtained by a second sensor through the filter bank to form a second set of sub-band output signals, the second sensor placed a distance from the first sensor, computing cross-correlation data between the first set of sub-band output signals and the second set of sub-band output signals, and applying a time delay determination function to the cross-correlation to determine a time delay estimation.

Time delay estimation

A method for presentation interaction. The method includes, receiving by a computer system an indication of a manual selection of a region proximate to an audience member of an audience wherein the indication is received via an interaction with a displayed image of the audience. The method also includes utilizing a microphone array communicatively coupled with a beam-forming component of the computer system to focus audio pickup from the region proximate to the audience member in response to receiving the indication. The method also includes displaying an enhanced image of the region proximate to the audience member using the computer system in response to receiving the indication.

Support for audience interaction in presentations

Systems and methods for performing sound source localization are provided. In one aspect, a method for locating a sound source using a computing device subdivides a space into subregions. The method then computes a sound source power for each of subregions and determines which of the sound source energies is the largest. When the volume of the subregion is less than a threshold volume, the method outputs the subregion having the largest sound source power. Otherwise, the stages of partitioning, computing, and determining the subregion having the largest sound source power is repeated.

Systems and methods for performing sound source localization

Embodiments of the present invention disclose a system and method for distributed meeting capture. According to one embodiment, the system includes a plurality of personal devices configured to capture video data and audio data associated with at least one operating user. A media hub includes a plurality of I/O ports and is configured to receive video and audio data from the plurality of personal devices. In addition, the media hub is configured to collect the video data and/or audio data from the plurality of personal devices and output at least one audio-visual data stream for facilitating video conferencing over a network.

System and method for distributed meeting capture

Disclosed herein are multimedia-conferencing systems and methods enabling local participants to hear remote participants from the direction the remote participants are rendered on a display. In one aspect, a method includes a computing device receives a remote participant's image and sound information collected at a remote site. The remote participant's image is rendered on a display at a local site. When the local participant is in close proximity to the display, sounds generated by the remote participant are played over stereo loudspeakers so that the local participant perceives the sounds as emanating from the remote participant's location rendered on the display.

Methods and systems for close proximity spatial audio rendering

Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.

Methods and systems for blind dereverberation

Various embodiments of the present invention are directed to systems and methods for multimodal object localization using one or more depth sensors and two or more microphones. In one aspect, a method comprises capturing three-dimensional images of a region of space wherein the object is located. The images comprise three-dimensional depth sensor observations. The method collects ambient audio generated by the object, providing acoustic observation regarding the ambient audio time difference of arrival at the audio sensors. The method determines a coordinate location of the object corresponding to the maximum of a joint probability distribution characterizing the probability of the acoustic observations emanating from each coordinate location in the region of space and the probability of each coordinate location in the region of space given depth sensor observations.

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Mountain View, CA, United States of America

Bowon Lee