Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.
Abstract:
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.