Cisco Technology, Inc.

Finisar Corporation

Tor A Sundsbarm

Feng Bao

Fei Yang

Subrahmanyam Venkata Kunapuli

David William Nolan Robison

Philip Roland Graham

Joseph T Friel

Richard Thayer Wales

Jian Zou

Stephen H Strong

David Michael Sanguinet

Xiangyu Bao

David Paul Wunsch Desrosiers

Olin Daniel Williford, Jr

Paul J Randall

In one example, a headset obtains a first audio signal including a user audio signal from a first microphone on the headset and a second audio signal including the user audio signal from a second microphone on the headset. The headset derives a first candidate signal from the first audio signal and a second candidate signal from the second audio signal. Based on the first audio signal and the second audio signal, the headset determines that a mechanical touch noise is present in one of the first audio signal and the second audio signal. In response to determining that the mechanical touch noise is present in one of the first audio signal and the second audio signal, the headset selects an output audio signal from a plurality of candidate signals including the first candidate signal and the second candidate signal. Headset provides the output audio signal to a receiver device.

Mechanical touch noise control

In one example, a headset obtains, from a first microphone on the headset, a first audio signal including a user audio signal and an anisotropic background audio signal. The headset obtains, from a second microphone on the headset, a second audio signal including the user audio signal and the anisotropic background audio signal. The headset extracts, from the first audio signal and the second audio signal, using a first adaptive filter, a reference audio signal including the anisotropic background audio signal. Based on the reference signal, the headset cancels, using a second adaptive filter, the anisotropic background audio signal from a third audio signal derived from the first and second audio signals to produce an output audio signal. The headset provides the output audio signal to a receiver device.

Anisotropic background audio signal control

A device including an array of bidirectional microphones optimizes the echo rejection of the bidirectional microphones. The microphone array receives audio from an audio source and generates audio signals from each of the bidirectional microphones. The device forms audio beams from combinations of the audio signals generated from the microphone array. Each audio beam captures audio from either its positive polarity zone or its negative polarity zone. The device determines a direction of the audio source and selects a perpendicular audio beam pair based on the direction of the audio source. The perpendicular audio beam pair includes a primary audio beam aimed toward the direction of the audio source and a secondary beam perpendicular to the primary audio beam. The device generates an output signal by combining the primary audio beam with the secondary audio beam based on polarity zone the audio is captured for each audio beam.

Beamforming and gainsharing mixing of small circular array of bidirectional microphones

A controller for the conference session generates a speaker signal for speakers in a conference room. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller estimates a relative location of the mobile device within the conference room. The controller dynamically selects as audio output corresponding to the mobile device location either the echo cancelled microphone signal from the mobile device or an echo cancelled array microphone signal associated with the relative location of the mobile device.

Acoustic echo cancellation for audio system with bring your own devices (BYOD)

Acoustic echo cancellation is improved by receiving a speaker signal that is used to produce audio in a room, and receiving audio signals that capture audio from an array of microphones in the room, including an acoustic echo from the speakers. To cancel the acoustic echo, one adaptive filter is associated with a corresponding subspace in the room. Each of the audio signals is assigned to at least one of the adaptive filters, and a set of coefficients is iteratively determined for each of the adaptive filters. The coefficients for an adaptive filter are determined by selecting each of the audio signals assigned to that adaptive filter and adapting the filter to remove an acoustic echo from each of the selected audio signals. At each iteration, a different audio signal is selected from the audio signals assigned to the adaptive filter in order to determine the set of coefficients.

Acoustic echo cancellation for microphone array with dynamically changing beam forming

A processing system can include tracking microphone array(s), audio-tracking circuitry configured to detect a location of audio sources from audio signals from the array(s), and processing circuitry. The processing circuitry can be configured to: identify a first microphone that has a strongest signal strength; estimate a location of an active speaker based on at least an output of the audio-tracking circuitry; determine whether a second microphone for the active speaker is affected by an acoustic obstacle based on the location of the active speaker and a location of the first microphone that has the strongest signal strength; estimate attenuation for microphones based on a comparison of actual signal strengths of the microphones with estimated signal strengths of the microphones that are estimated based on microphone signals of the second microphone for the active speaker; and modify the attenuation based on an estimated location of the acoustic obstacle.

Muting a sound source with an array of microphones

In one embodiment, a method includes obtaining a first signal from a first microphone, and determining when the first signal is indicative of activity on a first surface. The method also includes controlling a camera to focus on the first surface when it is determined that the first signal indicates the activity on the first surface. In such an embodiment, the first microphone and the camera may be part of a collaboration system, and the first surface may be a surface of a whiteboard.

Whiteboard activity noise detection system for immersive telepresence systems

A controller for the conference session receives at least one audio signal to generate a speaker signal. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller removes acoustic echo from the array microphone signals, and estimates a relative location of the mobile device. The controller dynamically selects as audio output corresponding to the mobile device location either (a) the array microphone signal, or (b) the echo cancelled microphone signal from the mobile device.

Methods and systems that compensate for display latency when separate speakers are used during video conferencing. A method includes determining whether speakers, which are not controlled by a display, are to be used in connection with a video conferencing session. The method further includes sending, to the display, data that causes the display to generate a predetermined pattern, capturing imagery of the effects of the predetermined pattern shown on the display, calculating a latency of the display based on a difference in a time the data was sent and the imagery of the effects is received, and storing a value of the latency of the display in a device that enables the video conferencing. When the speakers, which are not controlled by the display are selected, are selected, the audio portion of the video conferencing session is redirected to those speakers, but delayed for an amount of time substantially equivalent to the value of the latency of the display. In this way audio and video synchronization may be improved.

Automatic display latency measurement system for video conferencing

A system for synchronizing data packet collection and transmission on multiple segments of a local area network (&ldquo;LAN&rdquo;) or a wide area network (&ldquo;WAN&rdquo;) using a common clock to generate time stamps placed on the data packets by all peripheral network devices. The common clock is located on a network analyzer device which acts as the &ldquo;master&rdquo; to other &ldquo;slave&rdquo; peripheral network devices which are driven by the common clock and coupled to the master device in a master-slave configuration. The system also synchronizes the initialization of data packet transmission and/or collection on multiple peripheral network devices by using a common industry standard architecture (&ldquo;ISA&rdquo;) address for all devices involved in the data transmission and/or collection.

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San Jose, CA, United States of America

Tor A Sundsbarm