Abstract: A microphone assembly includes a base that includes a first surface, a second surface and a third surface. The first surface is configured to connect the base to a mounting location in an external environment. The second surface is positioned parallel to the first surface with the third surface positioned therebetween. The third surface defines a curved shape to direct air flow that contacts the third surface in a direction away from the first surface. The microphone assembly also includes a cap disposed over and separated from the base by a gap. The cap includes a domed portion that has a convex shape that curves away from the second surface of the base. The microphone assembly also includes a microphone array that includes a plurality of microphones. The microphone array is disposed within the cap and is configured to receive acoustic signals from the external environment through the gap.

Abstract: Presently described are systems and methods for identifying a mute/sound attribute of an audio sample set having first-channel samples and second-channel samples. An embodiment takes the form of a method that includes: (i) receiving the audio sample set; (ii) for each first-channel sample, determining a first max amplitude and a first min amplitude; (iii) calculating a first span based on a first function of the first max amplitude and the first min amplitude; (iv) for each second-channel sample, determining a second max amplitude and a second min amplitude; (v) calculating a second span based on a function of the second max amplitude and the second min amplitude; and (vi) identifying the audio sample set as having a sound attribute if both the first span and the second span are greater than a min-volume threshold.

Abstract: A device includes a receiver and a decoder. The receiver is configured to receive bitstream parameters corresponding to at least an encoded mid signal. The decoder is configured to generate a synthesized mid signal based on the bitstream parameters. The decoder is also configured to generate a synthesized side signal selectively based on the bitstream parameters in response to determining whether the bitstream parameters correspond to an encoded side signal.

Abstract: A method for determining an utterance direction includes: executing a first calculation process that includes determining an utterance direction region based on a phase difference of each given frequency between a first frequency signal and a second frequency signal inputted to a first and second voice input section respectively, and performing a process of calculating a first phase difference deviation degree indicating the degree by which the phase difference of each given frequency is deviated from the utterance direction region; executing a second calculation process for calculating a second phase difference deviation degree from the first phase difference deviation degree of the plurality of frames; and executing an utterance direction determination process for determining that a user is uttering a voice to the first voice input section and the second voice input section when the second phase difference deviation degree is equal to or smaller than a first threshold value.

Abstract: Systems and methods for capturing audio which can be used in applications such as virtual reality, augmented reality, and mixed reality systems. Some systems may include a plurality of distributed monitoring devices in an environment, each having a microphone and a location tracking unit. The system can capture audio signals while also capturing location tracking signals which indicate the locations of the monitoring devices over time during capture of the audio signals. The system can generate a representation of at least a portion of a sound wave field in the environment based on the audio signals and the location tracking signals. The system may also determine one or more acoustic properties of the environment based on the audio signals and the location tracking signals.

Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.

Abstract: A device for measuring wind noise comprises at least a first microphone and a processor. A first signal and a second signal are obtained from the at least one microphone, the first and second signals reflecting a common acoustic input, and the first and second signals being at least one of temporally distinct and spatially distinct. The first signal is processed to determine a first distribution of the samples of the first signal. The second signal is processed to determine a second distribution of the samples of the second signal. From a difference between the first distribution and the second distribution a scalar non-binary metric reflecting an intensity of wind noise present in the first and second signals is derived, and output.

Abstract: An earphone provided with a charging and/or data wire, having an earphone portion and a wire portion; one end of the earphone portion is fixedly provided with a wire socket; the wire portion has a wire for charging and/or data transmission; one end of the wire is fixedly provided with an earphone plug that matches with the wire socket and is connectable thereto; another end of the wire is fixedly provided with a wire plug.

Abstract: An electronic device includes a sound input interface, and a controller configured to be capable of recognizing sound that is input to the sound input interface. The controller is configured to, once recognizing predetermined sound other than voice of a user, inform, if information included in the predetermined sound matches schedule information of the user, information including the matched schedule information.

Abstract: A method for processing signals, a terminal device, and a non-transitory computer-readable storage medium are provided. The method includes the following. A sound signal of external environment is recorded via a microphone of a headphone when the headphone is in a playing state. Feature audio in the sound signal is identified and reminding information corresponding to the feature audio is acquired. Inquire of a user whether recorded sound signal is critical according to the reminding information in response to the headphone being paused. An input operation of the user is detected and the sound signal is processed according to the input operation of the user.

Abstract: The present invention extends to methods, systems, and computer program products for managing audio output through an intermediary. In some embodiments, an audio controller emulates a direct connection between an audio source device and an audio output device. In other embodiments, audio content local to an audio controller is combined with other audio content passing through the audio controller on its way from an audio source device to an audio output device. In additional embodiments, an audio output device is locked to an audio controller. The lock survives power cycling and soft resets of the audio output device. A special hard reset can be used to release the audio output device.

Abstract: Some disclosed systems may include a microphone system having two or more microphones, an interface system and a control system. In some examples, the control system may be capable of receiving, via the interface system, audio data from two or more microphones of the microphone system, of determining a gesture location based, at least in part, on the audio data and of controlling one or more settings of the system based on the gesture location.

Abstract: The present disclosure involves a method for automating media audio volume based on the physical motion of a mobile computing device. In one embodiment, the method includes detecting motion of the mobile device based on input motion data from a motion detection mechanism, determining whether the detected motion of the mobile device exceeds a configured motion threshold, based on the determining, causing a logical state change to transition into a automated volume state, and after the logical state change to transition to the automated volume state, automatically configuring the audio volume of audio output provided by a mobile computing device based on configured volume and fade parameters.

Abstract: A feedforward ambient noise reduction arrangement includes, within a housing, a loudspeaker device for directing sound energy into an ear of a listener. Disposed externally of the housing, and positioned to sense ambient noise on its way to the listener's ear, are plural microphone devices capable of converting the sensed ambient noise into electrical signals for application to the loudspeaker to generate an acoustic signal opposing the ambient noise. Importantly, the overall arrangement is such that the acoustic signal is generated by said loudspeaker means in substantial time alignment with the arrival of said ambient noise at the listener's ear.

Abstract: According to an exemplary aspect of the present invention, there is provided an apparatus comprising: at least one loudspeaker element, at least one processing core, at least one memory including computer program code, the at least one memory and the computer program configured to, with the at least one processing core, cause the apparatus to produce sound via the loudspeaker element, wherein the at least one processing core is configured to adjust the sound according to criteria stored in the at least one memory, wherein the criteria comprise a wide bandwidth roll-off using at least one parametric shelving filter.

Abstract: The disclosed technologies calibrate audio output based on a computer-based hearing evaluation. In an embodiment, the disclosed technologies perform operations that include playing an electronic file that contains segments of digital music that have different playback attributes, wherein playback attribute include at least one of volume data, frequency data, left/right data; during the playing of the electronic file, recording timestamp data and user account data that are associated with a user input; mapping the timestamp data to a set of playback attributes of a segment of the digital music; determining calibration data based on the timestamp data and the set of playback attributes; storing the calibration data in a user profile that is associated with the user account data; when different digital music is played, automatically calibrating the playing of the different digital music based on the user profile that is associated with the user account data.

Abstract: A method of operating a hearing aid system, comprising a hearing aid (200), adapted for detection of congestion of a sound output of the hearing aid. The invention also relates to hearing aid systems capable of carrying out such a method.

Abstract: An electronic device including an earphone device is provided. The earphone device includes a shell, a speaker, a first microphone device, a memory circuit and a controller. The memory circuit stores multiple parameter sets. The first microphone device receives a first sound. The first microphone device generates first data based on the first sound. The controller compares the first data with the parameter sets of the memory circuit and determines which one of the parameter sets corresponds to the first data based on the frequency parameters and the volume parameters. The controller generates second data based on the adjustment parameters of the one of the parameter sets, and the speaker generates a second sound based on the second data. The first sound generates a third sound in the shell, and the phase of the second sound is substantially opposite to the phase of the third sound.

Abstract: Disclosed is an active-noise control device including two passive attenuation earphones each provided with an external microphone, an internal microphone and a loudspeaker. The control device includes a first processing chain having a feedforward filter, a feedback filter and a reconstruction module. The control device includes a second processing chain having a sound source identification module, said second processing chain being implemented in parallel with the first processing chain, and being suitable for parameterizing the first processing chain.

Abstract: Boomless-microphones are described for a wireless helmet communicator with siren signal detection and classification capabilities. An acoustic component receives an audio signal and comprises a left acoustic sensor and a right acoustic sensor. The left acoustic sensor is mountable or attachable to the surface of a left wall of a helmet and the right acoustic sensor is mountable or attachable to the surface of a right wall. A speaker component can generate an echoless audio signal via signal inversion of the audio signal, outputs to a left speaker mountable or attachable to a left ear area of the helmet and a right speaker mountable or attachable to a right ear area of the helmet. A signal enhancement component can increase an intensity of the first audio signal associated with an emergency siren based on a determined proximity of an emitting emergency vehicle or emergency object to the device.