Embodiments relate generally to an active hearing protection device comprising a communication device operable to provide active hearing protection to a wearer; a set of earmuffs connected to the communication device, wherein each earmuff comprises an indicator mechanism operable to indicate to the communication device when the earmuff is inserted into a user's ear, a processor operable to receive information from the set of earmuffs and the indicator mechanisms of the earmuffs, and operable to automatically power the active hearing protection device on or off based on the information received from the indicator mechanism(s). When it is determined that both earmuffs are inserted into the user's ears, the active hearing protection device is powered on, and when it is determined that at least one of the earmuffs is not inserted into the user's ear(s), the active hearing protection device is powered off.

A method and system for daisy chaining tournament audio controllers, where the method comprises, in a headset coupled to a first audio controller, the first audio controller being in a daisy chain of audio controllers: receiving a chat signal from a second audio controller in the daisy chain of audio controllers, receiving a microphone signal from a microphone in the headset, summing the chat signal with the microphone signal, communicating the summed signal to a third audio controllers in the daisy chain, and communicating the chat signal to the headset. The microphone signal may be removed from the summed chat signal and microphone signal by adding a second microphone signal 180 degrees out of phase with the microphone signal. The chat signal may be summed with the microphone signal at an amplitude set by a user of the headset after the removal of the microphone signal.

A system for controlling a sound dose of a listener is presented. The system comprises a cloud server operatively connected to a remote storage configured to store a plurality of audio profiles, each comprising sound pressure data and a hearing profile associated with specific users of the system. The system further comprises one or more audio playback arrangements comprising a transducer, a controller, and a microphone arranged to measure a current sound pressure experienced by the listener. The cloud server and/or the controller of said one or more audio playback arrangements comprises controlling circuitry configured to, based on sound pressure data, a hearing profile of the listener and a current sound pressure experienced by the listener, cause estimating of a current estimated time period describing an estimated time until the sound dose of the listener exceeds a sound dose threshold. If the current estimated time period is below a dose period associated with the sound dose threshold, an updated sound pressure is calculated which describes an estimated time until the sound dose of the listener exceeds the sound dose threshold, meets or exceeds the dose period.

A noise-canceling device includes a processing circuit configured to detect vibrational noise sound waves near a listener's ear using a vibration sensor, generate a vibrational noise-canceling signal, and control operation of a speaker to provide a desired sound signal and the vibrational noise-canceling signal to at least partially cancel the vibrational noise sound waves.

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.

An audio system having low latency includes a digital audio processor as well as sensor inputs coupled to the processor. The sensor inputs may be microphone inputs. The audio processor operates at the same frequency as the sensor inputs, which is typically much higher than an audio signal provided to the audio processor. In some aspects the audio processor operates as a noise cancellation processor and does not include an audio input.

A method and computer program product for dynamically balancing the active noise cancellation value at a first and second earphone interface of a headphone device. The method enables determination of a first active noise cancellation value of an anti-noise output signal at a first interface. The method determines a second active noise cancellation value of the anti-noise output signal at a second interface. A comparison is made between the first active noise cancellation value and the second active noise cancellation value. In response to the first active noise cancellation value and the second active noise cancellation value being outside of a predetermined margin of each other, at least one adjustable parameter associated with the anti-noise output signal is dynamically adjusted to balance active noise cancellation values at the first and second headphone nterface.

An apparatus including at least one processor and at least one memory including computer program code configured to: when the apparatus is in a voice phone call mode of sound capture, output a signal from a first microphone as a primary speech signal, and select one or more second microphones to output a noise reduction signal, where the apparatus comprises at least two of the second microphones, and where selection of the one or more second microphones is based at least partially upon a determined use of the apparatus being either a handportable or an integrated handsfree use; and/or when the apparatus is in a multi-media mode of sound capture select one or more of the microphones to output audio signals based upon both a determined type of sound capture feature being used by the apparatus and based upon a determined orientation of the apparatus.

Systems and methods for ambient noise mitigation as a network service are provided. In some embodiments, an ambient noise mitigation server establishes at least one low latency network slice for at least one UE coupled to a radio access network. The ambient noise mitigation server generates a cancelation signal based on ambient sound mitigation data received by the radio access network, the ambient sound mitigation data including acoustic sensor data representing an ambient sound signal. The cancelation signal is generated to comprise a phase shift with respect to the ambient sound signal computed at least in part as a function of a location of the at least one UE, and causes at least one acoustic emitter to emit an acoustic cancelation signal based on the cancelation signal. In some embodiments, the phase shift may be adjusted by controlling a latency characteristic of the low latency network slice.

Disclosed is a headset including at least one in-ear microphone; at least one out-ear microphone; a control unit including an in-ear signal processing module for extracting low frequency components from a signal sensed through the in-ear microphone, an out-ear signal processing module for extracting high frequency components from a signal sensed through the out-ear microphone, and a mixing module for mixing the extracted low frequency components and high frequency components and outputting the mixed signal; and a communication unit for transmitting the signal output from the mixing module of the control unit to an external device, and sound quality can be improved by removing noises in a voice signal of a user transferred through the headset and creating a signal close to the voice of the user.