Cisco Technology, Inc.

Feng Bao

Tor A Sundsbarm

Subrahmanyam Venkata Kunapuli

Fei Yang

David William Nolan Robison

Jian Zou

David Michael Sanguinet

Xiangyu Bao

Kevin Lee Hughes

J William Mauchly

Stephen Lee Ijams

Victor Manuel Sanchez

Faisal Haq

David Paul Wunsch Desrosiers

Olin Daniel Williford, Jr

Disclosed herein are techniques for determining a wearing position of a boomless headset. An earpiece of the boomless headset can include at least one local talker (LT) microphone and a reference microphone. The LT microphone(s) are disposed substantially in a first end of the earpiece closest to a mouth of a LT when the LT wears the earpiece. The reference microphone is disposed substantially in a second end of the earpiece, furthest from the mouth of the LT when the LT wears the earpiece. A signal strength measurement (SSM) for a local talker audio signal to the LT microphone(s) and a SSM for a signal to the reference microphone are obtained. Signal processing logic can determine whether the earpiece is worn at an incorrect ear based on whether a difference between the SSM for the LT microphone(s) and the SSM for the reference microphone is below a predetermined threshold.

Wearing position detection of boomless headset

Presented herein is an audio endpoint for telecommunication operations with increased echo rejection. According to one example, the audio endpoint includes a housing body, an upper speaker assembly, a lower speaker assembly, and at least one microphone assembly. The upper speaker assembly is disposed near a top portion of the housing body and has an effective frequency range above a first frequency. The lower speaker assembly is disposed near a bottom portion of the housing body and has an effective frequency range below a second frequency. The microphone assembly includes a first microphone element and a second microphone element. The first microphone element is above the second microphone element so that they are vertically aligned. The first microphone element has an effective frequency range below the first frequency and the second microphone element has an effective frequency range above the second frequency.

Microphone assembly for echo rejection in audio endpoints

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

Presented herein is an audio endpoint for telecommunication operations, sometimes referred to herein as a 'telecommunications audio endpoint' or, more, simply as an 'audio endpoint.' According to at least one example, the audio endpoint presented herein includes a base, a speaker, a speaker waveguide, a microphone waveguide, and two or more microphones. The base is configured to engage a support surface (i.e., a table) and the speaker is configured to emit sounds (i.e., fire) in a direction of the base. The speaker waveguide is disposed between the speaker and the microphone waveguide, while the microphone waveguide is disposed between the speaker waveguide and the base. The two or more microphones are disposed within the microphone waveguide and are proximate to the base. In general, the speaker waveguide is configured to guide sounds output by the speaker in general radially (outward) directions.

Telecommunications audio endpoints

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.

Clock synchronization for an acoustic echo canceller (AEC) with a speaker and a microphone connected over a digital link may be provided. A clock difference may be estimated by analyzing the speaker signal and the microphone signal in the digital domain. The clock synchronization may be combined in both hardware and software. This synchronization may be performed in two stages, first with coarse synchronization in hardware, then fine synchronization in software with, for example, a re-sampler.

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Sunnyvale, CA, United States of America

Feng Bao